ras: VG uses RAS message to interact information with GK Server to map the called phone number to the IP address of peer voice gateway. It is used only in the networking configuration that uses GK (gatekeeper) to provide voice IP services.

Description

Use the address command to configure the voice routing policy to the peer voice gateway.

Use the undo address command to cancel the voice routing policy that has been configured.

By default, no routing policy to the peer VG is configured.

This command is used to configure the network address for the VolP voice entity. The system supports the following three VolP routing policies at present.

l Static routing policy: Find the IP address of destination voice gateway in static mode according to the address ip ip-address command.

l Dynamic routing policy: The VG and GK Server exchange RAS information after using the address ras command. GK will dynamically send back the peer voice gateway address that matches the called number to the VG.

l Static and dynamic integrated routing policy: Two voice entities have the same match-template configuration (namely the same called number) and have been configured with address ras and address ip ip-address commands respectively. Therefore, the system will first select the route according to the configured routing policy. If the select entity fails, the system will select another entity to complete the call.

Related command: match-template, priority, and select-rule.

Example

# Set the IP address of the called voice gateway of a VoIP voice entity (the number is 12345) to 10.1.1.2.

[VG-voice-dial-entity1] match-template 12345

[VG-voice-dial-entity1] address ip 10.1.1.2

1.1.2 area

Syntax

area { asia |north-america | custom | europe }

undo area

View

FXO voice subscriber-line view

Parameter

asia: The busy tone type of the PBX connected to this subscriber-line is of Asia standard.

north-america: The busy tone type of the PBX connected to this subscriber-line is of North America standard.

custom: The busy tone type of the PBX connected to this subscriber-line is defined by the users.

europe: The busy tone type of the PBX connected to this subscriber-line is of Europe standard.

Description

Use the area command to configure the type of busy tone detection for FXO voice subscriber-line.

Use the undo area command to restore the default value.

By default, europe busy tone type standard is set.

This command is used only for 2-wire loop trunk subscriber-line FXO.

When this subscriber-line is connected to a common user line of a program-controlled PBX, if the user on the PBX side hooks on first, only by detecting the busy tone can the VG know the user on-hooking operation. Since different PBXs execute different prompt tone schemes, there exist different frequency spectrum characteristics. This command is used to set the frequency spectrum characteristic used by the VG to detect the existence of the busy tone.

Example

# Use north-america standard to detect the existence of the busy tone on voice subscriber-line 4.

[VG-voice-line4] area north-america

1.1.3 area-id (voice entity)

Syntax

area-id string

undo area-id

View

VoIP voice entity view

Parameter

string: Area ID, a character string consisting of numbers from 0 to 9 and “#”.

Description

Use the area-id command to configure the area ID of VG.

Use the undo area-id command to cancel the specified area ID.

By default, no area ID of VG is configured.

The voice area ID is set in VoIP voice entity view and will be automatically added to the beginning of called numbers when establishing calls.

Related command: match-template and entity.

Example

# Configure the VoIP voice entity 101 with the area ID 6#.

[VG-voice-dial-entity101] area-id 6#

1.1.4 busytone-t-th

Syntax

busytone-t-th time-threshold

undo busytone-t-th

View

FXO voice subscriber-line view

Parameter

time-threshold: Number of the busy tone detection cycles. The more cycles, the longer detection time. It ranges from 2 to 12 and defaults to 2, that is, the device goes onhook after it detects busy tone data of two consecutive cycles.

Description

Use the busytone-t-th command to configure the busy tone detection threshold.

Use the undo busytone-t-th command to restore the default value of the busytone detection threshold.

The actual busy tone data does not always match the configured busy tone parameter, and excessive mismatch may result in on-hook failure or improper on-hook. Adjustment of the DSP busy tone detection time can make the detection more accurate.

Increasing the time for busy tone detection improves the accuracy of busy tone detection, which helps avoid, to a certain extent, improper on-hooks caused by busy tone data inaccuracy; however, this may increase the probability of on-hook failures.

Note that before you configure a threshold of busy tone detection, you must test it fully making sure that on-hook operation can be done properly.

Example

# Set the busy tone detection threshold to 3.

[VG-voice-line0] busytone-t-th 3

1.1.5 caller-permit

Syntax

caller-permit permit-num

undo caller-permit { permit-num | all }

View

Voice entity view

Parameter

all: All callers.

permit-num: Calling numbers that are permitted to call in, in the format of { [ + ] string [ $ ]| $ }. The largest length of the string is 31. The symbols are described in the following:

l +: Appears at the beginning of a calling number to indicate that the number is E.164-compliant.

l $: As the last character to indicate the end of the number. That means the entire calling number must match all the characters before “$” in the string. If there is only “$” in the string, the calling number can be empty.

l string: A string composed of any characters of “0123456789#*.!+%[]()-”. The meanings of the characters are described in the following table:

Table 1-1 Meanings of the characters in string

Character	Meaning
0 — 9	Each digit, among 0 and 9, represents a digit.
# and *	Each represents a valid digit.
.	Wildcard, which can match any digit of a valid number. For example, “555. . . .” matches any number string that starts with 555 and has four additional characters.
-	Connector, used to connect two values (it is preceded by the smaller one and followed by the larger one) to express a range. For example, “1-9” represents the range of 1 to 9 (including 1 and 9).
[ ]	Represents a character selection range. It can be used together with “!” “%”, or “+”. For example, “[235-9]” represents only one single character of “2” or “3”, or between “5” and “9” can be matched.
( )	Represents a group of characters. For example, “(086)” represents the character string “086”. This pair is usually used together with “!” “%” or “+”. For example, “(086)!010” can match the two character strings “086010” and “010”.
!	Specifies that the preceding sub-expression can be absent or present once. For example, “(010)!12345678” can match both “12345678” and “01012345678”.
+	Specifies that the preceding sub-expression can be present one or more times. However, when it appears at the beginning of a whole number, it means that this number is an E.164 number. In this case, the “+” character neither represents any specific number nor means number repetition. For example, “(1)9876(54)+” can match “987654”, “98765454”, “9876545454” and so on. “(2)+110022” means that 110022 is an E.164 number.
%	Specifies that the preceding sub-expression can be absent or present multiple times. For example, “9876(54)%” can match “9876”, “987654”, “98765454”, “9876545454” and so on.

& Note:

l The sub-expression in front of "!”, “%”, and “+” are not to be matched accurately. They are handled similar to the wildcard “.”. Moreover, these symbols cannot be used alone. There must be a valid digit or digits in front of them.

l You can to use “[ ]” inside “( )”, namely, in the form of “( [ ] )”. Other forms, such as “[ [ ] ]” and “[ ( ) ]” are illegal.

l “-” can only be used inside “[ ]”, and it only connects the same type of characters, such as “0-9”. The formats like “0-A” are illegal.

Description

Use the caller-permit command to configure the calling numbers that are permitted to call in.

Use the undo caller-permit command to delete the calling numbers that are permitted to call in.

By default, no calling number is configured. That means there is no limitation on calling numbers.

You can configure 32 calling numbers for a voice entity at most. If you only use “$”, empty calling numbers are permitted to call in.

Related command: match-template.

Example

# Configure voice entity 2 to permit 660268 or empty calling numbers to call in.

[VG-voice-dial-entity2] caller-permit 660268$

[VG-voice-dial-entity2] caller-permit $

# Configure voice entity 2 to permit the calling numbers starting with 20 to call in.

[VG-voice-dial-entity2] caller-permit 20

1.1.6 cid display

Syntax

cid display

undo cid display

View

Voice subscriber-line view

Parameter

None

Description

Use the cid display command to enable the calling number display.

Use the undo cid display command to disable the calling number display.

By default, it enables the calling number display.

This command only applies to FXS voice subscriber-lines.

When the FXS is the called party, it can display the calling number between the first and the second rings. If the FXS interface is disabled to send calling numbers, it sends to the called telephone the “P” character which is received from the IP side. Thus, the called telephone is disabled to display calling numbers.

Example

# Display calling number identification on voice subscriber-line 1.

[VG-voice-line1] cid display

1.1.7 cid enable

Syntax

cid enable

undo cid enable

View

voice subscriber-line view

Parameter

None

Description

Use the cid enable command to enable the calling identification (CID).

Use the undo cid enable command to disable the CID.

By default, CID is enabled on the FXO interface.

This command only applies to FXO voice subscriber-lines.

The FXO can receive the modulation data of the calling number through an analog line between the first and second rings, possess it with FSK demodulation, and finally send the calling number data to IP side.

Example

# Enable calling number sending on voice subscriber-line 1.

[VG-voice-line1] cid enable

1.1.8 cid send

Syntax

cid send

undo cid send

View

Voice subscriber-line

Parameter

None

Description

Use the cid send command to enable the voice gateway to send calling numbers to the IP side.

Use the undo cid send command to disable the voice gateway to send calling numbers to the IP side.

By default, calling numbers are sent to the IP side.

This command applies to FXS and FXO voice subscriber-lines.

Example

# Disable the voice gateway to send calling numbers to the IP side on voice subscriber-line 4 (FXO).

[VG-voice-line4] cid send

# Disable the voice gateway to send calling numbers to the IP side on voice subscriber-line 0 (FXS).

[VG-voice-line0] undo cid send

1.1.9 cid type

Syntax

cid type { complex | simple }

View

Voice subscriber-line view

Parameter

complex: Configure the current subscriber-line to send the calling number information in complex format.

simple: Configure the current subscriber-line to send the calling number information in simple format.

Description

Use the cid type command to configure the current message format adopted by the subscriber-line to send the calling information.

So far, a message can be in either complex or simple format. If the remote device can only support either format, you should keep the local format in consistency with the remote format.

This command is only available to FXS and FXO voice subscriber-line.

By default, the complex format is used.

Example

# Send the calling number in simple format on voice subscriber line 1.

[VG-voice-line1] cid type simple

1.1.10 cng-on

Syntax

cng-on

undo cng-on

View

Voice subscriber-line view

Parameter

None

Description

Use the cng-on command to enable comfort noise function.

Use the undo cng-on command to disable the comfort noise function,

By default, comfort noise setting is enabled.

This command is applicable to subscriber-lines created on the FXO, FXS, and E1VI interfaces. When the silence detecting function on a corresponding voice entity is enabled, some background noise can be generated by using the command to fill the toneless intervals during a conversation. If no comfort noise is generated, the toneless intervals during a conversation will cause the interlocutors uncomfortable.

Related command: subscriber-line and vad-on.

Example

# Disable comfort noise function on subscriber line 1.

[VG-voice-line1] undo cng-on

1.1.11 compression

Syntax

compression { 1st-level | 2nd-level | 3rd-level | 4th-level } { g711alaw | g711ulaw | g723r53 | g723r63 | g729a | g729r8 }

undo compression { 1st-level | 2nd-level | 3rd-level | 4th-level }

View

Voice entity view

Parameter

1st-level: Indicate the first selected voice compression method.

2nd-level: Indicate the second selected voice compression method.

3rd-level: Indicate the third selected voice compression method.

4th-level: Indicate the fourth selected voice compression method.

g711alaw: Indicate the A-law coding mode of G.711 (defining the pulse code modulation technology), the bandwidth is 64 kbps, usually adopted by Europe.

g711ulaw: Indicate the μ-law coding mode of G.711, the bandwidth is 64 kbps, usually adopted by countries of the North America and Japan, etc.

g723r53: Indicate G.723.1 Annex A coding mode (dual rate voice coding of multimedia communication which is based on multi-pulse multi-quantitative level and the Algebraic-Code-Excited Linear-Prediction technologies), the bandwidth is 5.3 kbps. Currently, the VG 20-16, VG 20-32, and VG 21-08 do not support this mode.

g723r63: Indicate G.723.1 Annex A coding mode, the bandwidth is 6.3 kbps. Currently, the VG 20-16, VG 20-32 and VG 21-08 do not support this mode.

g729a: Indicate Appendix A mode of G.729, the bandwidth is 8 kbps.

g729r8: Indicate G.729 (the voice coding technology using conjugate Algebraic-Code-Excited Linear-Prediction) coding mode, the bandwidth is 8 kbps.

Description

Use the compression command to configure the voice compression mode according to priority level.

Use the undo compression command to restore the default value.

By default, g729r8 coding mode is set.

Using the compression command, you can set voice compression method of different priority levels.

g711alaw and g711ulaw coding provide high-quality voice transmission, but occupy greater bandwidth.

g723r53 and g723r63 coding provide silence suppression technology and comfort noise, the relatively higher speed output is based on multi-pulse multi-quantitative level technology and provides relatively higher voice quality to certain extent, and the relatively lower speed output is based on the Algebraic-Code-Excited Linear-Prediction technology and provides greater flexibility for application.

The voice quality provided by the g729r8 and g729a coding is similar to the ADPCM of 32 kbps, having the quality of a toll, and also featuring low bandwidth, lesser event delay and medium processing complexity, hence it has a wide field of application.

The following table describes the relation between codec algorithms and bandwidth. Usually, 8000Hz is adopted to collect voice samples. The bandwidth without compression is 64kbps, and it is compressed using the ITU-T G series codec algorithms. The table also shows the compression ratio.

Table 1-2 Relation between algorithms and bandwidth

Codec algorithm	Bandwidth	Voice quality
G.711 (A-law and m-law)	64kbps (without compression)	Best
G.729	8kbps	Good
G.723 r63	6.3kbps	Fair
G.723 r53	5.3kbps	Fair

Actual network bandwidth is related to packet assembly interval and network structure. The longer the packet assembly interval is, the closer the network bandwidth is to the media stream bandwidth. More headers consume more bandwidth. Longer packet assembly interval results in longer fixed coding latency.

The following tables show the relevant packet assembly parameters without IPHC compression, including packet assembly interval, bytes coded in a time unit, and network bandwidth, etc. Thus, you can choose a suitable codec algorithm according to idle and busy status of the line and network situations more conveniently.

Table 1-3 G.711 algorithm (A-law and m-law)

Packet assembly interval	Bytes coded in a time unit	Packet length IP	Network bandwidth IP	Packet length IP+PPP	Network bandwidth IP+PPP	Coding latency
20ms	160	200	80Kbps	206	82.8Kbps	20ms
30ms	240	280	74.7Kbps	286	76.3Kbps	30ms
G.711 algorithm (A-law and m-law): media stream bandwidth 64kbps, minimum packet assembly interval 20ms.

Table 1-4 G.729 algorithm

Packet assembly interval	Bytes coded in a time unit	Packet length IP	Network bandwidth IP	Packet length IP+PPP	Network bandwidth IP+PPP	Coding latency
20ms	20	60	24Kbps	66	26.4Kbps	20ms
30ms	30	70	18.7Kbps	76	20.3Kbps	30ms
G.729 algorithm: media stream bandwidth 8Kbps, minimum packet assembly interval 20ms.

Table 1-5 G.723 r63 algorithm

Packet assembly interval	Bytes coded in a time unit	Packet length IP	Network bandwidth IP	Packet length IP+PPP	Network bandwidth IP+PPP	Coding latency
30ms	24	64	16.8Kbps	70	18.4Kbps	30ms
60ms	48	88	11.6Kbps	94	12.3Kbps	60ms
90ms	72	112	9.8Kbps	118	10.3Kbps	90ms
G.723 r63 algorithm: media stream bandwidth 6.3Kbps, minimum packet assembly interval 30ms.

Table 1-6 G.723 r53 algorithm

Packet assembly interval	Bytes coded in a time unit	Packet length IP	Network bandwidth IP	Packet length IP+PPP	Network bandwidth IP+PPP	Coding latency
30ms	20	60	15.9Kbps	66	17.5Kbps	30ms
60ms	40	80	10.6Kbps	86	11.4Kbps	60ms
90ms	60	100	8.8Kbps	106	9.3Kbps	90ms
G.723 r53 algorithm: media stream bandwidth 5.3Kbps, minimum packet assembly interval 30ms.

& Note:

l Packet assembly interval is the duration to encapsulate information into a voice packet.

l Bytes coded in a time unit = packet assembly interval X media stream bandwidth.

l Packet length (IP) = IP header + RTP header + UDP header + voice information length = 20+12+8+data

l Packet length (IP+PPP) = PPP header + IP header + RTP header + UDP header + voice information length = 6+20+12+8+data

l Network bandwidth = 64kbps X packet length / bytes coded in a time unit

Since IPHC compression is affected significantly by network stability, it cannot achieve high efficiency unless line is of high quality, network is very stable, and packet loss does not occur or seldom occurs. When the network is unstable, IPHC efficiency drops drastically. With best IPHC performance, IP (RTP) header can be compressed to 2 bytes. If PPP header is compressed at the same time, a great deal of media stream bandwidth can be saved. The following table shows the best IPHC compression efficiency of codec algorithms with packet assembly interval of 30ms.

Table 1-7 Compression efficiency of IPHC+PPP header

Coding modes	Bytes coded in a time unit	Before compression		After Compression
Coding modes	Bytes coded in a time unit	Packet length IP+PPP	Network bandwidth IP+PPP	Packet length IP+PPP	Network bandwidth IP+PPP
G.729	30	76	20.3Kbps	34	9.1Kbps
G.723r63	24	70	18.4Kbps	28	7.4Kbps
G.723r53	20	66	17.5Kbps	24	6.4Kbps

Only when there is an intersection (a compression mode recognized by both parties) in the voice compression modes owned by the two communication parties can the two parties establish normal communication with each other. If there is no consistency in the compression modes set for the equipment at the two ends connected to each other, or there is no common compression method, the calling will fail.

Example

# Configure to select g723r53 compression method first, then to select the g729r8 compression method.

[VG-voice-dial-entity1] compression 1st-level g723r53

[VG-voice-dial-entity1] compression 2nd-level g729r8

1.1.12 cptone

Syntax

cptone { locale | cs } [ { type | all } amplitude value ]

undo cptone [ { locale | cs } { type | all } amplitude ]

View

Voice subscriber-line view

Parameter

locale: Number that represents a country or an area. Set this argument to specify the call progress tone (cptone) mode for the current voice subscriber line to the mode of a specific country or area among the 62 countries and areas supported currently.

cs: Represents "custom”, that is, sets the cptone mode for the current voice subscriber line to customized mode.

type: Cptone type. Currently the supported cptone types include the dial tone, special dial tone, busy tone, congestion tone, ringback tone and waiting tone.

all: Represents all types of cptone.

amplitude value: Specifies amplitude of each type of cptone. The value ranges from 200 to 2000, with 1000 as the default for the busy tone and congestion tone, and 600 as the default for other cptone types.

Table 1-8 Types of cptone

Type	Description
dial-tone	Dial tone
special-dial-tone	Special dial tone
congestion-tone	Congestion tone
busy-tone	Busy tone
ringback-tone	Ringback tone
waiting-tone	Waiting tone

Table 1-9 Modes of cptone

Mode	Country/area
AR	Argentina
AU	Australia
AT	Austria
BE	Belgium
BR	Brazil
BG	Bulgaria
CA	Canada
CL	Chile
CN	China
HR	Croatia
CU	Cuba
CS	Custom
CY	Cyprus
CZ	Czech Republic
DK	Denmark
EG	Egypt
FI	Finland
FR	France
DE	Germany
GH	Ghana
GR	Greece
HK	Hong Kong China
HU	Hungary
IS	Iceland
IN	India
ID	Indonesia
IR	Iran
IE	Ireland
IL	Israel
IT	Italy
JP	Japan
JO	Jordan
KE	Kenya
KR	Korea Republic
LB	Lebanon
LU	Luxembourg
MY	Malaysia
MX	Mexico
NP	Nepal
NL	Netherlands
NZ	New Zealand
NG	Nigeria
NO	Norway
PK	Pakistan
PA	Panama
PH	Philippines
PL	Poland
PT	Portugal
RU	Russian Federation
SA	Saudi Arabia
SG	Singapore
SK	Slovakia
SI	Slovenia
ZA	South Africa
ES	Spain
SE	Sweden
CH	Switzerland
TH	Thailand
TR	Turkey
GB	United Kingdom
US	United States
UY	Uruguay
ZW	Zimbabwe

Description

Use the cptone command to specify the cptone mode for the current voice subscriber line as the mode of a specific country or area or as the customized mode.

Use the undo cptone command to restore the default cptone mode.

By default, the cptone mode is CN.

& Note:

The cptone locale command takes effect on all voice ports on the board where the current voice subscriber line is attached.

Example

# Set the cptone mode to US.

[VG-voice-line1/0/0] cptone us

1.1.13 debugging voice data-flow

Syntax

debugging voice data-flow { all | detail | error | fax [ error ] | jitter [ error ] | receive | send | vpp }

View

Any view

Parameter

all: Enable the information debugging output switch of all voice data processes.

detail: Enable the information debugging output switch of detailed information of voice packets.

error: Enable the error information debugging output switch of voice data stream.

fax [ error ]: Enable the information debugging output switch of fax data stream, the parameter error is used to enable the error information debugging switch of fax data stream.

jitter [ error ]: Enable the information debugging output switch of Jitter Buffer processing in the data stream, the parameter error is used to enable the error information debugging switch of Jitter Buffer processing.

receive: Enable the information debugging output switch on the receiving side of the data stream.

send: Enable information debugging output switch on send side of the data stream.

vpp: Enable the information debugging output switch of the data stream of VPP software module.

Description

Use the debugging voice data-flow command to enable information output switch of voice data process.

Use the undo debugging voice data-flow command to disable the debugging switch.

Example

# Enable the information debugging output switch of all voice data processes.

[VG] debugging voice data-flow all

1.1.14 debugging voice dpl

Syntax

debugging voice dpl { all | error | general }

View

Any view

Parameter

all: Enables all debugging of voice dial program.

error: Enables error debugging of voice dial program.

general: Enables general debugging of voice dial program.

Description

Use the debugging voice dpl command to enable the debugging of voice dial program.

Use the undo debugging voice dpl command to disable the debugging.

Example

# Enable debugging of voice dial program.

[VG] debugging voice dpl all

1.1.15 debugging voice h225

Syntax

debugging voice h225 { asn1 | event }

View

Any view

Parameter

asn1: The output information is the information related to negotiation messages.

event: The output information is the information related to negotiation events.

Description

Use the debugging voice h225 command to enable the H.225.0 debugging of negotiation message or event.

Use the undo debugging voice h225 command to disable the debugging.

Example

# Enable H.225 negotiation events debugging.

[VG] debugging voice h225 event

1.1.16 debugging voice h245

Syntax

debugging voice h245 event

View

Any view

Parameter

event: The output information is the information related to negotiation events.

Description

Use the debugging voice h245 command to enable H.245 debugging for negotiation events.

Use the undo debugging voice h245 command to disable H.245 debugging for negotiation events.

Example

# Enable H.245 negotiation events debugging.

[VG] debugging voice h245 event

1.1.17 debugging voice ipp

Syntax

debugging voice ipp { all | error | rtp-rtcp | socket | timer | vcc | vpp }

View

Any view

Parameter

all: Enable all the debugging functions on IPP module.

error: Enable error debugging on IPP module.

rtp-rtcp: Enable RTP/RTCP information debugging.

socket: Enable Socket information debugging.

timer: Enable timer information debugging.

vcc: Enable VCC message receiving and sending debugging.

vpp: Enable VCC message receiving and sending debugging.

Description

Use the debugging voice ipp command to enable H.323 protocol stack module debugging.

Use the undo debugging voice ipp command to disable this debugging function.

Example

# Enable all the debugging functions of the H.323 protocol stack IPP module.

[VG] debugging voice ipp all

1.1.18 debugging voice rcv

Syntax

debugging voice rcv { all | cc | error | timer | vas | vcc | vpp }

View

Any view

Parameter

all: Enable all the debugging of the RCV module.

cc: Enable the debugging of call control of the RCV module.

error: Enable the debugging of connection failure caused by the RCV module.

timer: Enable the debugging on the timer operation of the RCV module.

vas: Enable the debugging between the RCV module and the bottom layer VAS module.

vcc: Enable the debugging between the RCV module and the upper layer VCC module.

vpp: Enable the debugging between the RCV module and the bottom layer VPP module.

Description

Use the debugging voice rcv command to enable the debugging of the RCV module.

Use the undo debugging voice rcv command to disable the debugging.

Example

# Enable all the debugging of the RCV module.

[VG] debugging voice rcv all

1.1.19 debugging voice vas

Syntax

debugging voice vas { all | buffer | channel channel-number | cid | command | dsp | error | fax | rcv | receive | send }

View

Any view

Parameter

all: Enable all the output debugging of the VAS module.

buffer: Enable the debugging of the buffer area for the VAS module to transmit commands to the DSP module.

channel channel-number: Enable the debugging of the specified output channel of the VAS module.

cid: VAS CID debugging

command: Enable the debugging between the VAS module and the command buffer area.

dsp: Enable the debugging between the VAS module and the bottom layer DSP module.

error: Enable the output debugging of the connection failure caused by the VAS module.

fax: Enable the fax debugging of the VAS module.

rcv: Enable the debugging between the VAS module and the RCV module.

receive: Enable the debugging for the VAS module to receive data.

send: Enable the debugging for the VAS module to transmit data.

Description

Use the debugging voice vas command to enable the VAS module debugging.

Use the undo debugging voice vas command to disable the debugging.

The debugging voice vas channel channel-number command is independent of the command debugging voice vas all. Use the debugging voice vas channel channel-number command to enable the debugging for the specified channel. For the detailed debugging information, combine this command with other debugging information. Use the debugging voice vas all command to enable the debugging for all the VAS except the channel.

The undo debugging voice vas all command is not completely independent of the undo debugging voice vas channel command. Using the undo debugging voice vas all command will enable the undo debugging voice vas channel command while the undo debugging voice vas channel command only disable the debugging for the specified channel.

Example

# Enable all the debugging of the VAS module.

[VG] debugging voice vas all

1.1.20 debugging voice vmib

Syntax

debugging voice vmib { aaaclient | all | analogif | callactive | callhistory | dialcontrol | digitalif | error | general | gkclient | h323statistic | voiceif }

undo debugging voice vmib { aaaclient | all | analogif | callactive | callhistory | dialcontrol | digitalif | error | general | gkclient | h323statistic | voiceif }

View

Any view

Parameter

aaaclient: AAA client debugging.

all: Voice MIB debugging.

analogif: Analog voice debugging.

callactive: Current calling debugging.

callhistory: Calling history record debugging.

dialcontrol: Dialing debugging.

digitalif: Digital voice debugging.

error: Voice MIB error debugging.

general: Voice MIB basic debugging.

gkclient: GK client debugging.

h323statistic: H.323 message accounting debugging.

voiceif: Voice debugging.

Description

Use the debugging voice vmib command to enable voice MIB debugging.

Use the undo debugging voice vmib command to disable the voice MIB debugging.

By default, no voice MIB is enabled.

Example

# Enable the H.323 message statistic debugging of voice MIB.

[VG] debugging voice vmib h323statistic

1.1.21 debugging voice vpp

Syntax

debugging voice vpp { all | codecm | error | ipp | rcv | timer | vas | vcc }

View

Any view

Parameter

all: Enable all the output debugging of the VPP module.

codecm: Enable the debugging between the VPP module and the bottom layer CODECM module.

error: Enable the output debugging of the connection failure of the VPP module.

ipp: Enable the debugging between the VPP module and the upper layer IPP module.

rcv: Enable the debugging between the VPP module and the RCV module

timer: Enable the output information debugging on the timer operation of the VPP module.

vas: Enable the debugging between the VPP module and the VAS module.

vcc: Enable the debugging between the VPP module and the VCC module

Description

Use the debugging voice vpp command to enable the debugging of the VPP module.

Use the undo debugging voice vpp command to disable the debugging switch.

Example

# Enable all the output debugging of the VPP module.

[VG] debugging voice vpp all

1.1.22 default entity compression

Syntax

default entity compression { 1st-level | 2nd-level | 3rd-level | 4th-level } { g711alaw | g711ulaw | g723r53 | g723r63 | g729a | g729r8 }

undo default entity compression { 1st-level | 2nd-level | 3rd-level | 4th-level }

View

Voice dial program view

Parameter

Refer to the compression command.

Description

Use the default entity compression command to globally configure the mode of coding and decoding as the default value.

Use the undo default entity compression command to restore the fixed value (i.e.: g729r8 code mode) in the system as the default value.

By default, the mode of coding and decoding is g729r8 coding mode.

The default entity compression command can be used to globally configure the default value of the voice coding and decoding. In this case, all the voice entities and newly created voice entities on this VG which have not been configured with this function will inherit this configuration.

Related command: compression.

currently, the VG 20-16, VG 20-32 and VG 21-08 do not support g723r53 and g723r63.

Example

# Adopt the g723r53 coding and decoding mode as the first selection globally.

[VG-voice-dial] default entity compression 1st-level g723r53

1.1.23 default entity fast-connect

Syntax

default entity fast-connect

undo default entity fast-connect

View

Voice dial program view

Parameter

None

Description

Use the default entity fast-connect command to globally enable the fast connection mode as the default setting.

Use the undo default entity fast-connect command to restore the default value.

By default, the default value globally configured to the fast connection mode is disabled.

Example

# Configure to enable the default value globally configured to the fast connection mode.

[VG-voice-dial] default entity fast-connect

1.1.24 default entity payload-size

Syntax

default entity payload-size { g711 | g723 | g729 } time-length

undo default entity payload-size { g711 | g723 | g729 }

View

Voice dial policy view

Parameter

g711: g711 coding/decoding. It can be g711alaw or g711ulaw.

g723: g723 coding/decoding. It can be g723r53 or g723r63.

g729: g729 coding/decoding. It can be g729r8 or g729a.

time-length: Time length of the DSP packets with the specified coding/decoding mode. See Table 1-10 for its value range.

Description

Use the default entity payload-size command to set the default time length for DSP to assemble a packet, that is, the default time length of each voice packet.

Use the undo default entity payload-size command to restore the default time length of voice packets that is used by default.

By default, it takes 20 milliseconds for DSP to assemble a packet with g711, and 30 milliseconds for DSP to assemble a packet with g723, or g729. Currently, VG20-16, VG 20-32 and VG 21-08 do not support g723.

For each coding/decoding mode, the maximum value range of the time length varies with voice gateways. If you specify a time length beyond the range, the default applies. And as such, when you find out that a configured time length does not work, you should first check that its value is in the allowed range with the adopted coding/decoding mode and on the device.

The following table lists the default time length ranges and rules for DSP to assemble a packet.

Table 1-10 Time length ranges of voice packets

G711

G723

G729

VG 20-32

VG 20-16

VG 21-08

An integral multiple of 10, in the range of 20 to 30

An integral multiple of 10, in the range of 20 to 180

VG 10-40

VG 10-41

An integral multiple of 30, in the range of 30 to 180

Related command: payload-size.

Example

# Set the default time length for DSP to assemble a packet with g729a to 180 milliseconds.

[VG-voice-dial ] default entity payload-size g729 180

1.1.25 default entity vad-on

Syntax

default entity vad-on

undo default entity vad-on

View

Voice dial program view

Parameter

None

Description

Use the default entity vad-on command to globally configure enabling silence detection as the default value.

Use the undo default entity vad-on command to restore the fixed value (i.e.: disabling the silence detection) to be the default value.

By default, the silence detection is disabled.

The default entity vad-on command is used to globally configure enabling silence detection as the default value. In this case, all the voice entities and newly created voice entities on this VG, which have not been configured with this function, will inherit this configuration.

Related command: vad-on.

Example

# Enable the silence detection globally.

[VG-voice-dial] default entity vad-on

1.1.26 default subscriber-line dtmf gain

Syntax

default subscriber-line dtmf gain { hf | lf } value

undo default subscriber-line dtmf gain { hf | lf }

View

Voice view

Parameter

hf: Gain of high frequency signals, ranging from –6.0 dB to +14.0 dB, with one-decimal-digit precision. The default value is 0 dB.

lf: Gain of low frequency signals, ranging from –6.0 dB to +14.0 dB, with one-decimal-digit precision. The default value is 0 dB.

Description

Use the default subscriber-line dtmf gain command to configure the gain of high and low DTMF signals of all voice subscriber lines. Use the undo default subscriber-line dtmf gain command to restore the default value.

This command is applicable to the FXO and FXS interfaces.

Related command: dtmf gain.

Example

# Configure the gain of high and low DTMF signals of all voice subscriber lines to –2.0 dB.

[VG-voice] default subscriber-line dtmf gain hf -2.0

[VG-voice] default subscriber-line dtmf gain lf -2.0

1.1.27 default subscriber-line receive gain

Syntax

default subscriber-line receive gain value

undo default subscriber-line receive gain

View

Voice view

Parameter

value: Voice input gain. The value is a real number ranging from –6.0 dB to +14.0 dB, with one-decimal-digit precision. The default value is 0 dB.

Description

Use the default subscriber-line receive gain command to configure the voice input gain of all voice subscriber lines. Use the undo default subscriber-line receive gain command to restore the default value.

When the input line needs low power for voice signals, this command can be used to increase the voice input attenuation, thereby meeting the line requirements.

Related commands: receive gain, transmit gain, and default subscriber-line transmit gain.

Example

# Configure all voice input gains to –2.0 dB.

[VG-voice] default subscriber-line receive gain -2.0

1.1.28 default subscriber-line transmit gain

Syntax

default subscriber-line transmit gain value

undo default subscriber-line transmit gain

View

Voice view

Parameter

value: Voice output gain. The value is a real number ranging from –6.0 dB to +14.0 dB, with one-decimal-digit precision. The default value is 0 dB.

Description

Use the default subscriber-line transmit gain command to configure the voice output gain of all voice subscriber lines. Use the undo default subscriber-line transmit gain command to restore the default value.

When the output line needs low power for voice signals, this command can be used to increase the voice output attenuation, thereby meeting the line requirements.

Related commands: receive gain, transmit gain, and default subscriber-line receive gain.

Example

# Configure all voice output gains to –2.0 dB.

[VG-voice]default subscriber-line transmit gain -2.0

1.1.29 delay

Syntax

delay { dtmf | dtmf-interval } milliseconds

delay start-dial seconds

undo delay { dtmf | dtmf-interval start-dial }

View

FXO voice subscriber-line view

Parameter

dtmf milliseconds: Duration of the DTMF signals of the FXO interface, ranging from 50 to 500 milliseconds and the default value is 120 milliseconds.

dtmf-interval milliseconds: DTMF signal interval between FXO interface, ranging from 50 to 500 milliseconds and the default value is 120 milliseconds.

start-dial seconds: Delayed time for the FXO interface to start dial the number in the range from 0 to 10 seconds. The default value is 1 second.

Description

Use the delay command to configure the relevant time parameters on an FXO interface.

Use the undo delay command to restore the default values of these time parameters.

All the commands listed above are used for configuring the device of a calling party and hence are only useful for the calling party.

Related command: timer.

Example

# Set the duration of DTMF signals on the FXO interface to 150 milliseconds.

[VG-voice-line1] delay dtmf 150

1.1.30 description (voice entity)

Syntax

description string

undo description

View

Voice entity view

Parameter

string: Voice entity description string, with length ranging from 1 to 64 characters.

Description

Use the description command to configure a voice entity description string.

Use the undo description command to delete the voice entity description string.

You can make a description of the voice entity by using the description command. This operation will not affect the performance of voice entities at all. You can view its information when executing the display command.

Example

# Identify voice entity 10 with local-entity 10.

[VG-voice-dial-entity10] description local-entity10

1.1.31 description (voice subscriber-line)

Syntax

description string

undo description

View

Voice subscriber-line view

Parameter

string: Interface description character string, and its value range is 1 to 64 characters.

Description

Use the description command to configure an interface description character string.

Use the undo description command to cancel the interface description of character string.

By default, no interface description is configured.

This command is applicable to the FXO, FXS and E1VI interfaces.

With the description command, make a description of the voice subscriber-line connection. This operation will not have any influence on the voice entities. The configuration information can be seen only when the display command is executed.

Example

# Identify subscriber line 1 as connected to laboratory lab_1.

[VG-voice-line1] description lab_1

1.1.32 dial-prefix

Syntax

dial-prefix string

undo dial-prefix

View

Voice entity view

Parameter

string: Prefix code, a number of fixed length. The string is composed of any characters from “0123456789, #*”, with the largest length of 31 characters. The meanings of the characters are shown in the following table.

Table 1-11 Meanings of the characters in string

Character	Meaning
0-9	Numbers from 0 to 9. Each means a digit.
,	A comma means a pause of 500ms. It can be placed on any position of the number.
# and *	Each means a valid digit.

Description

Use the dial-prefix command to configure the prefix of the telephone number dialed by the voice entity.

Use the undo dial-prefix command to cancel the prefix of the telephone number dialed by the voice entity.

This command only applies to the configuration of POTS voice entity. And the dial-prefix command only applies to the FXO interface. Whether to send a second stage dialing tone is determined by the configuration of the PBX connected to the VG.

When a VG receives a voice call, it makes a comparison between the number configured in the match-template of its own POTS voice entity and the number received, and selects one POTS voice entity to continue the call processing.

If send-number is set to its default value (truncate), the VG will remove from the called number the string that matches the match-template beginning from the left. If the dial-prefix command is configured, the prefix will be added in front of the rest of the called number. The VG will initiate a call according to the new string. Supposing that the called number is 0102222, the called number template of the voice entity that is configured by match-template is 010…., and the dial prefix is 0, then “010” that accurately matches the template will be removed, and the rest part “2222” will be added a prefix “0”. The VG initiates a call to the new number 02222.

If the number with the added prefix contains more than 31 digits, only the first 31 characters will be sent.

Related command: match-template and send-number.

Example

# Use “0” as a prefix.

[VG-voice-dial-entity1] dial-prefix 0

1.1.33 dial-program

Syntax

dial-program

View

Voice view

Parameter

None

Description

Use the dial-program command to enter the voice dial program view.

Use the quit command to return to the voice view.

In the dial program view, conduct the configuration and policies related to numbers such as mapping number to voice device, number extension, and number matching.

Example

# Enter the voice dial program view.

[VG-voice] dial-program

1.1.34 display voice call-history-record line

Syntax

display voice call-history-record line line-number

View

Any view.

Parameter

line-number: Subscriber line number.

Description

Use the display voice call-history-record line command to view the call information of a subscriber line.

Usually, save the called numbers of the last 10 calls, without time limit.

The command is used to display the subscriber-line type, line status and the latest incoming/outgoing calls of a specified subscriber line, as well as the called numbers of the last 10 calls originated by the local end. In the case of less than ten calls, the actual number of calls will be displayed.

& Note:

The specified voice subscriber line number must not be greater than the number of available voice subscriber lines; otherwise, the input will be invalid.

Example

# Display the call information of a subscriber line.

[VG] display voice call-history-record line 0

Subscriber-line 0 type FXS POTS , Line state is opened

start outgoing call 72 times, 48 success

receive incoming call 9 times, 6 success

the latest 10 called number is:

%1% called number 900

%2% called number 900

%3% called number 900

%4% called number 900

%5% called number 17912

%6% called number 17920

%7% called number 17920

%8% called number 17920

%9% called number 2186

%10% called number 2526

Table 1-12 Description on the fields of the display voice call-history-record line command

Item	Description
Subscriber-line	Index of subscriber-line
type	Interface type of subscriber-line
Line state	The line status of subscriber-line
start outgoing call	The total number of outgoing calls and the number of successful outgoing calls
receive incoming call	The total number of incoming calls and the number of successful incoming calls
the latest 10 called number	The latest 10 called number from this subscriber-line

1.1.35 display voice call-info

Syntax

display voice call-info { brief | detail | mark TAG }

View

Any view

Parameter

brief: Displays the call information table in brief.

detail: Displays the call information table in detail.

mark TAG: Displays the call information according to the mark number in the call information table. The TAG value range is 1 to 128.

Description

Use the display voice call-info command to display the call information table, including: channel number of the call, reference counter of all voice modules, module ID in use, list of the voice entities that can be selected by the current call, and the voice entity used by the current call.

Example

# Display the call information table of a certain time in brief:

[VG] display voice call-info brief

The information table for current calls in brief

! CALL ( 0): Channel <--> 0

Module ID <--> RCV VAS VCC IPP VPP

! CALL ( 1): Channel <--> 2

Module ID <--> RCV VAS VCC IPP VPP

End

# Display the call information table of a certain time in detail:

The information table for current calls in detail

**************** CALL 0 ***************

Channel number : 0

Reference counter : 5

Module check ID :

RCV VAS VCC IPP VPP

Current used voice entity : 3

Voice entities are offered :

**************** CALL 1 ***************

Channel number : 2

Reference counter : 5

Module check ID :

RCV VAS VCC IPP VPP

Current used voice entity : 3

Voice entities are offered :

End

# Display the record with the mark number of 1 in the call information table.

[VG] display voice call-info mark 1

**************** CALL 0 ***************

Channel number : 0

Reference counter : 5

Module check ID :

RCV VAS VCC IPP VPP

Current used voice entity : 3

Voice entities are offered :

1.1.36 display voice default

Syntax

display voice default all

View

Any view

Parameter

None

Description

Use the display voice default command to view the current default values and the system-fixed default values.

This command can be used to display the current default values and system-fixed default values for voice and fax. For example, both the current default setting and the system-fixed default setting are to truncate called numbers, the carrier transmission energy level of GW defaults to -10 (the system-fixed default value is 15) and so on.

Example

# Display the current default values and the system-fixed default values.

[VG] display voice default all

Default entity fax ecm off(system: off)

Default entity fax protocol t38(system: t38)

Default entity fax redundancy hb-redundancy 0(system: 0)

Default entity fax redundancy lb-redundancy 0(system: 0)

Default entity fax level -15(system: -15)

Default entity fax local-train threshold 10(system: 10)

Default entity fax baudrate voice(system: voice)

Default entity fax nsf-on off(system: off)

Default entity fax support-mode rtp(system: rtp)

Default entity fax train-mode ppp(system: ppp)

Default entity compression 1st-level g729r8(system: g729r8)

Default entity compression 2nd-level g711alaw(system: g711alaw)

Default entity compression 3rd-level g711ulaw(system: g711ulaw)

Default entity compression 4th-level g723r53(system: g723r53)

Default entity voice activity detect off(system: off)

Default entity VoIP calling fast connect off(system: off)

Default entity payload-size g711 20 (system: 20)

Default entity payload-size g723 30 (system: 30)

Default entity payload-size g729 30 (system: 30)

Table 1-13 Description on the fields of the display default command

Item	Description
fax ecm	Use ECM for fax
fax protocol	Fax interoperability protocol
fax redundancy hb-redundancy	Number of high-bit redundancy packets (for the H.323-T.38 or T.38 fax protocol only)
fax redundancy lb-redundancy	Number of low-bit redundancy packets (for the H.323-T.38 or T.38 fax protocol only)
fax level	Carrier transmission energy level of GW
fax local-train threshold	Percentage of the fax local train threshold
fax baudrate	Highest fax rate
fax nsf-on	Negotiation mode of fax facilities
fax support-mode	Fax support mode
fax train-mode	Fax training mode
compression 1st-level	The first level voice codec method
compression 2nd-level	The second level voice codec method
compression 3rd-level	The third level voice codec method
compression 4th-level	The fourth level voice codec method
cancel-truncate	Cancel truncate called number on voice entity
vad-on	Voice activity detection
fast connect	Fast connect at the calling end of VoIP voice entity
payload-size g711	Voice entity packing duration (g711)
payload-size g723	Voice entity packing duration (g723)
payload-size g729	Voice entity packing duration (g729)

1.1.37 display voice entity

Syntax

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

View

Any view

Parameter

all: All voice entities.

pots: All POTS voice entities.

voip: All VoIP voice entities.

mark: Displays a voice entity.

entity-tag: Tag of the voice entity that is to be displayed, ranging from 1 to 2147483647.

Description

Use the display voice entity command to view the configuration information of voice entities of different types.

Usually, you can view the information of all the interfaces that are active in the VG and the global configuration by executing the display current-configuration command. But it will display a great deal of information. So if you just want to view the configuration information of voice entities, you can use the display voice entity command.

Example

# Display the configuration information of POTS voice entities.

[VG] display voice entity pots

Current configuration of entities

entity 66 pots

match-template 6600..

shutdown

compression 1st-level g711alaw

dial-prefix 6600

line 6

entity 67 pots

match-template 6600..

shutdown

compression 1st-level g711alaw

dial-prefix 6600

line 7

End

1.1.38 display voice ipp

Syntax

display voice ipp { ccb | statistic { all | h225 | h245 | ras | socket | timer | vcc | vpp } }

View

Any view

Parameter

ccb: displays the call control block information of the IPP module.

statistic: Displays the statistics information of IPP module.

all: Displays all the statistics information of the IPP module.

h225: Displays H225 message statistics information.

h245: Displays H245 message statistics information.

ras: Displays RAS message statistics information.

socket: Displays socket message statistics information.

timer: Displays timeout statistics information.

vcc; Displays the statistics information between the IPP module and the VCC module.

vpp: Displays the statistics information between the IPP module and the VPP module.

Description

Use the display voice ipp command to display the statistics information of the IPP module.

Example

# Display the H225 message statistics information of the IPP module.

[VG] display voice ipp statistic h225

statistics about H225 :

{

Send_Setup : 0

Send_CallProceeding : 0

Send_Alerting : 0

Send_Connect : 0

Send_ReleaseComplete : 0

Send_FacilityIndUserInput : 0

Send_FacilityTCSRequest : 0

Send_FacilityTCSAck : 0

Send_FacilityTCSReject : 0

Send_FacilityOLCRequest : 0

Send_FacilityOLCAck : 0

Send_FacilityOLCReject : 0

Send_FacilityMSDRequest : 0

Send_FacilityMSDAck : 0

Send_FacilityMSDReject : 0

Send_FacilityCLCRequest : 0

Send_FacilityCLCAck : 0

Send_FacilityStartH245 : 0

Send_Error : 0

Recv_Setup : 0

Recv_CallProceeding : 0

Recv_Alerting : 0

Recv_Connect : 0

Recv_ReleaseComplete : 0

Recv_Progress : 0

Recv_FacilityTCSRequest : 0

Recv_FacilityTCSAck : 0

Recv_FacilityTCSReject : 0

Recv_FacilityOLCRequest : 0

Recv_FacilityOLCAck : 0

Recv_FacilityOLCReject : 0

Recv_FacilityMSDRequest : 0

Recv_FacilityMSDAck : 0

Recv_FacilityMSDReject : 0

Recv_FacilityCLCRequest : 0

Recv_FacilityCLCAck : 0

Recv_Unknow : 0

}

1.1.39 display voice number-substitute

Syntax

display voice number-substitute [list-tag ]

View

Any view

Parameter

list-tag: Serial number of the number-substitute list, ranging from 1 to 2147483647.

Description

Use the display voice number-substitute command to view the configuration information of number-substitute lists. It can display the information of a certain list and all the lists.

Related command: number-substitute.

Example

# Display all the configured number-substitute lists.

[VG] display voice number-substitute

Current configuration of number substitute

************ NUMBER-SUBSTITUTE ************

List-tag : 1

First-rule : INDEX_INVALID

Dot-match : end-only

rule 0

Input-format : ^011408

Output-format : 1408

End

1.1.40 display voice rcv ccb

Syntax

display voice rcv ccb

View

Any view

Parameter

None

Description

Use the display voice rcv ccb command to view the information related to the call control block in the RCV module.

This command is used to display the information related to the incoming and outgoing call control block, the connection status of modules, the calling status, the caller numbers and the called numbers, etc.

Example

# Display the information related to the call control block in the RCV module.

[VG] display voice rcv ccb

RCV : CCB [ 1 ]

{

CallID : 0x0043

CallState : TALK

VCCID : 0x004f

VCCState : IPPS_CONNECTED

CCID : 0xffff

CCState : CCS_CONNECTED

VASID : 0xffff

CallType : OUTGOING

CallAttribute : 0x00000000

CallSignaling : 0x00000002

EncodeType : 0x0000001f

E1Slot : 0xffffffff

E1Port : 0xffffffff

TimeSlot : 0xffffffff

ChannelID : 0x00000003

VPUState : VS_CONNECTED

VCCTIMER : 0x00000000

CCTimer : 0x00000000

VPUTimer : 0x00000000

CCChanMsg : 0x00000000

E1ChanMsg : 0x00000000

CallerNumber : 111

CalledNumber : 660010

prev : 0x00000000

next : 0x01409500

}

RCV : CCB [ 2 ] {

CallID : 0x0042

CallState : TALK

VCCID : 0x004e

VCCState : IPPS_CONNECTED

CCID : 0xffff

CCState : CCS_CONNECTED

VASID : 0x0039

CallType : INCOMING

CallAttribute : 0x00000000

CallSignaling : 0x00000004

EncodeType : 0x00000009

E1Slot : 0xffffffff

E1Port : 0xffffffff

TimeSlot : 0xffffffff

ChannelID : 0x00000000

VPUState : VS_CONNECTED

VCCTimer : 0x00000000

CCTimer : 0x00000000

VPUTimer : 0x00000000

CCChanMsg : 0x00000000

E1ChanMsg : 0x00000000

CallerNumber : 111

CalledNumber : 660010

prev : 0x01409000

next : 0x00000000

}

Table 1-14 Description on the fields of the display voice rcv ccb command

Item	Description
CCB [ ]	Index of call control block
CallID	Flag or Identifier of the calling
VCCID	Index of VCC module in calling period
VCCState	Status of VCC module in calling period
CCID	Index of CC module in calling period
CCState	Status of CC module in calling period
VASID	Index of VAS module in calling period
CallType	Type of the calling
CallAttribute	Attribute of the calling
CallSignaling	Signaling of the calling
EncodeType	Voice compression method of the calling
E1Slot	Number of slot where E1V1 board is located for the calling
E1Port	Number of CE1/PRI interface for the calling
TimeSlot	Timeslot on the E1 of the calling
ChannelID	Identifier of logic channel of the calling
VPUState	Status of VPU of the calling
VCCTimer	Timer of VCC module in calling period
CCTimer	Timer of CC module in calling period
VPUTimer	Timer of VPU module in calling period
CCChanMsg	Pointer of CC channel message in calling period
E1ChanMsg	Pointer of E1VI channel message in calling period
CallerNumber	Calling number of the calling
CalledNumber	Called number of the calling
prev	Previous RCV call control block
next	Next RCV call control block

1.1.41 display voice rcv statistic

Syntax

display voice rcv statistic { all | call | error | proc | timer | vas | vcc | vpp }

View

Any view

Parameter

all: Display all the statistic information of the RCV module.

call: Display the calling statistic information in the RCV module.

error: Display all the statistic information on the failure of connection caused by the RCV module.

proc: Display the statistic information of process calls in the RCV module.

timer: Display all the statistic information on the timer operation in the RCV module.

vas: Display all the statistic information between the RCV module and the bottom layer VAS module.

vcc: Display the calling statistic information between RCV module and the upper layer VCC module.

vpp: Display all the statistic information between the RCV module and the bottom layer VPP module.

Description

Use the display voice rcv statistic command to view the statistic information of calling between the RCV module and the CC module, IPP module VAS module, etc.

The all parameter can enable you to display the total number of calling, the successful calling, and the failed calling of VAS-related modules in RCV module. Using other parameters such as IPP and VAS, you can display the statistic information in the corresponding modules.

Example

# Display the statistic information of calling between the RCV module and other modules.

[VG] display voice rcv statistic call

statistics between RCV and VCC :

{

Send_VCC_CCSetup : 0

Send_VCC_CCAlerting : 0

Send_VCC_CCConnect : 0

Send_VCC_CCRelease : 0

Recv_VCC_IPCallOut : 0

Recv_VCC_IPAlerting : 0

Recv_VCC_IPConnect : 0

Recv_VCC_IPCallProceeding : 0

Recv_VCC_IPRelease : 0

Recv_VCC_Unknow : 0

}

statistics between RCV and VPP :

{

Send_ConnectVPU : 0

Send_DisConnectVPU : 0

Recv_ConnectVPUAck_VCR_SUCCESS : 0

Recv_ConnectVPUAck_VCR_FAIL : 0

Recv_DisConnectVPUAck : 0

Recv_UnConnect : 0

Recv_VPP_Unknow : 0

}

statistics between RCV and CC :

{

Send_CONN_REQ : 0

Send_CONN_RES : 0

Send_DISC_REQ : 0

Send_DISC_RES : 0

Send_STATUS_REQ : 0

Recv_CONN_CFM : 0

Recv_CONN_IND : 0

Recv_DISC_CFM : 0

Recv_DISC_IND : 0

Recv_STATUS_IND_INFO_ALERT : 0

Recv_STATUS_IND_INFO_CHANNELID : 0

Recv_STATUS_IND_ELSE : 0

Recv_CC_Unknow : 0

}

statistics between RCV and R2 :

{

Send_R2_ConnectReqAck_SUCCESS : 0

Send_R2_ConnectReqAck_FAIL : 0

Send_R2_ActiveAck_SUCCESS : 0

Send_R2_ActiveAck_FAIL : 0

Send_R2_Onhook : 0

Send_R2_Offhook : 0

Send_R2_IPAlerting : 0

Recv_R2_ConnectReq : 0

Recv_R2_Active_TD_IN : 0

Recv_R2_Active_TD_OUT : 0

Recv_R2_Active_ELSE : 0

Recv_R2_Release : 0

Recv_R2_Alert_AP_ALERTING : 0

Recv_R2_Alert_ELSE : 0

Recv_R2_Unknow : 0

}

statistics between RCV and VAS :

{

Send_VPM_Offhook : 0

Send_VPM_ConnAck : 0

Send_VPM_Onhook : 0

Send_VPM_InAlert : 0

Recv_VPM_ConnPre : 0

Recv_VPM_ConnReq : 0

Recv_VPM_OutAlert : 0

Recv_VPM_Active : 0

Recv_VPM_Release : 0

Recv_VPM_OnhookAck : 0

Recv_VPM_PolarityReverse : 0

Recv_VPM_Unknow : 0

}

statistics between RCV and Timer :

{

Send_TIMEOUT_CONNECTVPU : 0

Send_TIMEOUT_DISCONNECTVPU : 0

Send_TIMEOUT_IPALERTING : 0

Send_TIMEOUT_IPCONNECT : 0

Send_TIMEOUT_STATUS_IND : 0

Send_TIMEOUT_CONN_CFM : 0

Send_TIMEOUT_DISC_CFM : 0

Send_TIMEOUT_STATUS_IND_ALERT : 0

Send_TIMEOUT_R2_ALERT : 0

Send_TIMEOUT_R2_ACTIVE : 0

Send_TIMEOUT_VPM_OUT_ALERT : 0

Send_TIMEOUT_VPM_ACTIVE : 0

Send_TIMEOUT_VPM_CONNREQ : 0

Send_TIMEOUT_VPM_ONHOOKACK : 0

Recv_TIMEOUT_CONNECTVPU : 0

Recv_TIMEOUT_DISCONNECTVPU : 0

Recv_TIMEOUT_IPALERTING : 0

Recv_TIMEOUT_IPCONNECT : 0

Recv_TIMEOUT_STATUS_IND : 0

Recv_TIMEOUT_CONN_CFM : 0

Recv_TIMEOUT_DISC_CFM : 0

Recv_TIMEOUT_STATUS_IND_ALERT : 0

Recv_TIMEOUT_R2_ALERT : 0

Recv_TIMEOUT_R2_ACTIVE : 0

Recv_TIMEOUT_VPM_OUT_ALERT : 0

Recv_TIMEOUT_VPM_ACTIVE : 0

Recv_TIMEOUT_VPM_CONNREQ : 0

Recv_TIMEOUT_VPM_ONHOOKACK : 0

Recv_TIMEOUT_Unknow : 0

}

statistics about RCV errors :

{

RCV_SEND_VCC_MSG_FAILED : 0

RCV_SEND_VPP_MSG_FAILED : 0

RCV_SEND_CC_MSG_FAILED : 0

RCV_SEND_R2_MSG_FAILED : 0

RCV_SEND_VPM_MSG_FAILED : 0

RCV_REQUEST_TIMER_FAILED : 0

RCV_CONNECT_VPU_TIMEOUT : 0

RCV_IPALERTING_TIMEOUT : 0

RCV_IPCONNECT_TIMEOUT : 0

RCV_STATUS_IND_TIMEOUT : 0

RCV_STATUS_IND_ALERT_TIMEOUT : 0

RCV_CC_CONN_CFM_TIMEOUT : 0

RCV_R2_ALERT_TIMEOUT : 0

RCV_R2_ACTIVE_TIMEOUT : 0

RCV_VPM_ALERT_TIMEOUT : 0

RCV_VPM_ACTIVE_TIMEOUT : 0

RCV_VPM_CONNREQ_TIMEOUT : 0

RCV_NO_AVAILABLE_LOGICAL_CHANNEL : 0

RCV_VPP_UNCONNECT_RELEASE : 0

RCV_REQUEST_VPU_CHANNEL_FAILED : 0

RCV_GET_SIGNAL_FAILED : 0

RCV_FIND_PORT_FAILED : 0

RCV_GET_SLOT_PORT_FAILED : 0

RCV_SELECT_IDLE_CHANNEL_FAILED : 0

RCV_RECV_MESSAGE_IN_WRONG_STATUS : 0

RCV_TIMEOUT_IN_WRONG_STATUS : 0

RCV_FIND_CCB_FAILED : 0

RCV_FIND_CCB_BY_VCCID_FAILED : 0

RCV_FIND_CCB_BY_CCID_FAILED : 0

RCV_FIND_CCB_BY_VPMID_FAILED : 0

RCV_FIND_CCB_BY_CHANNELID_FAILED : 0

RCV_GET_SENDNUM_BY_CALLEDNUM_FAILED : 0

RCV_OTHER_RELEASE : 0

}

statistics about RCV Procedure :

{

RCV_NewCCB : 0

RCV_DeleteCCB : 0

RCV_FindCCB : 0

RCV_FindCCBByCcID : 0

RCV_FindCCBByVccID : 0

RCV_FindCCBByVpmID : 0

RCV_FindCCBByChID : 0

RCV_SelectE1TimeSlot : 0

RCV_RelCall : 0

RCV_InitChannel : 0

}

statistics about RCV calls :

{

RCV_CC_ACTIVE_CALL : 0

RCV_CC_ACTIVE_CALL_SUCCEEDED : 0

RCV_CC_ACTIVE_CALL_FAILED : 0

RCV_CC_PASSIVE_CALL : 0

RCV_CC_PASSIVE_CALL_SUCCEEDED : 0

RCV_CC_PASSIVE_CALL_FAILED : 0

RCV_R2_ACTIVE_CALL : 0

RCV_R2_ACTIVE_CALL_SUCCEEDED : 0

RCV_R2_ACTIVE_CALL_FAILED : 0

RCV_R2_PASSIVE_CALL : 0

RCV_R2_PASSIVE_CALL_SUCCEEDED : 0

RCV_R2_PASSIVE_CALL_FAILED : 0

RCV_VPM_ACTIVE_CALL : 0

RCV_VPM_ACTIVE_CALL_SUCCEEDED : 0

RCV_VPM_ACTIVE_CALL_FAILED : 0

RCV_VPM_PASSIVE_CALL : 0

RCV_VPM_PASSIVE_CALL_SUCCEEDED : 0

RCV_VPM_PASSIVE_CALL_FAILED : 0

}

Table 1-15 Description on the fields of the display voice rcv statistic command

Item	Description
RCV_VAS_ACTIVE_CALL	Number of active call between self and VAS module
RCV_VAS_ACTIVE_CALL_SUCCEEDED	Successful number of active call between self and VAS module
RCV_VAS_ACTIVE_CALL_FAILED	Failing number of active call between self and VAS module
RCV_VAS_PASSIVE_CALL	Number of passive call between self and VAS module
RCV_VAS_PASSIVE_CALL_SUCCEEDED	Successful number of passive call between self and VAS module
RCV_VAS_PASSIVE_CALL_FAILED	Failing number of passive call between self and VAS module

1.1.42 display voice subscriber-line

Syntax

display voice subscriber-line line-number

View

Any view

Parameter

line-number: The subscriber line number.

Description

Use the display voice subscriber-line command to view the configuration information about the type, status, codec mode, input and output gains of the subscriber line.

Related command: subscriber-line.

Example

# Display the configuration information about the subscriber line.

[VG] display voice subscriber-line 1

Current information --- line: 1

Type = FXS POTS

Status = UP -- CH_IDLE

Coding =

Decoding =

CallerNum =

CalledNum =

Call-ID = 0

Call-Refer = 0

CNG = ON

EchoCancel = ON - 32 (ms)

Reset = 0

Position = Slot 1 Port 1

Pickup = FALSE

CID-Display = ENABLE

CID-Send = ENABLE

Gain(R&T) = 0 (db) - 0 (db)

T_FirstDial = 10 (s)

T_DialInter = 4 (s)

[VG] display voice subscriber-line 4

Current information --- line: 4

Type = LINE FXO

Status = UP-- CH_IDLE

Coding =

Decoding =

CallerNum =

CalledNum =

Call-ID = 0

Call-Refer = 0

CNG = ON

EchoCancel = ON - 32 (ms)

Reset = 0

Position = Slot 0 Port 4

CID-Send = ENABLE

CID-Receive = ENABLE

Gain(R&T) = 0 (db) - 0 (db)

T_FirstDial = 10 (s)

T_DialInter = 4 (s)

T_Predial = 1 (s)

T_DTMF = 120 (ms)

T_Interdigit= 120 (ms

Table 1-16 Description on the fields of the display voice subscriber-line command

Item	Description
line	Index of voice subscriber-line
Type	Type of voice subscriber-line
Status	Status of voice subscriber-line
Status information	Calling status of voice subscriber-line
Coding	Encoding protocol
Decoding protocol	Decoding protocol
CallerNum	Calling number of current call
CalledNum	Called number of current call
Call-ID	Identifier of current call
Call-Refer	Call reference of current call
CNG	Comfort noise configuration on the voice subscriber-line
EchoCancel	Echo cancellation on the voice subscriber-line
Reset	The reset times of the board associated with the voice subscriber-line is located
Position	The slot number and port number of the board associated with the voice subscriber-line
CID-Send	Display the calling number
CID-Receive	Display calling number about CID function
Gain(R&T)	Gain on the voice subscriber-line
T_DialInter	Timeout of wait digit dialed by user for subscriber-line
T_Predial	Delay time of pre-dial on subscriber-line
T_DTMF	Duration time of DTMF digit on subscriber-line
T_Interdigit	Interval time of DTMF digit on subscriber-line

1.1.43 display voice voip data-statistic

Syntax

display voice voip data-statistic [ channel channel- number | detail ]

View

Any view

Parameter

channel channel-number :Display statistic information of voice data on specified voice subscriber-line.

detail: Display detailed information of logical channel of voice.

Description

Use the display voice voip data-statistic command to view statistic information of voice data.

This command is used to display the following information: the time of successfully searching the voice table, the total number of received data packets, the time of searching the table in fast and common modes, the voice and fax information of the receive and transmit channels, and so on. When the detail keyword is provided, this command displays detailed information of receive and send channels (including detailed input/output statistic and jitter Buffer of information packets).

Related command: vqa data-statistic and reset voice voip data-statistic.

Example

# Display the statistic information of voice data.

[VG] display voice voip data-statistic

=== VoIP datagram summary ===

--------------------------------------------------------------------

[NET] SearchVoiceTableSuccess : 0

[NET] ReceiveDatagramTotal : 0

------------------------ Receive channel: 000 ----------------------

[NET] ReceiveDatagramTotal : 0

[COM] DiscardDatagramTotal : 0

[NET] ReceiveRtpDatagram : 0

[NET] ReceiveRtcpDatagram : 0

[COM] AddReceiveTable : 3

[COM] ClearReceiveTable : 0

[COM] FreeReceiveBuffer : 0

[COM] TrsmitDone : 462

VOICE DATA INFORMATION:

[NET] ReceiveDataTotal : 0

[NET] NormalProcessData : 0

FAX DATA INFORMATION:

[NET] ReceiveDataTotal : 0

[NET] NormalProcessData : 0

------------------------ Send channel: 000 ----------------------

COMMON INFORMATION:

[VPP] LinkReceiveDataTotal : 484

[VPP] LinkReceiveEmptyData : 0

[COM] AddTableTimes : 0

[COM] ClearTableTimes : 0

VOICE DATA INFORMATION:

[FSD] FSSend_ReceiveDataTotal : 483

[COM] DiscardDataTotal : 0

[NET] NormalSendDataTotal : 0

[NET] SendToIPDataTotal : 0

[LOC] SendToLocalDataTotal : 484

FAX DATA INFORMATION:

[FSD] FSSend_RcvFaxDataTotal : 0

[FDF] DiscardDataTotal : 0

[NET] SendToIPDataTotal : 0

[LOC] SendToLocalDataTotal : 0

Table 1-17 Description on the fields of the display voice voip data-statistic command

Item	Description
SearchVoiceTableSuccess	Successful times of searching voice table
ReceiveDatagramTotal (summary)	Number of total packets received
ReceiveDatagramTotal(channel)	Number of total packets received in channel
DiscardDatagramTotal	Number of total packets dropped in channel
ReceiveRtpDatagram	Number of total RTP packets received in channel
ReceiveRtcpDatagram	Number of total RTCP packets received in channel
AddReceiveTable	Times of adding receiving table
ClearReceiveTable	Times of resetting receiving table
FreeReceiveBuffer	Times of clearing buffer on board
TrsmitDone	Times of freeing transmit buffer
ReceiveDataTotal(voice)	Number of total voice data received in channel
NormalProcessData(voice)	Number of total voice data processed normally in channel
ReceiveDataTotal(fax)	Number of total fax data received in channel
NormalProcessData(fax)	Number of total fax data processed normally in channel
LinkReceiveDataTotal	Number of total local data received in channel
LinkReceiveEmptyData	Number of total null local data received in channel
AddTableTimes	Times of adding sending table
ClearTableTimes	Times of resetting sending table
FSSend_ReceiveDataTotal(voice)	Number of total voice data fast received or sent
DiscardDataTotal(voice)	Number of total voice data dropped
NormalSendDataTotal	Number of total voice data sent normally
SendToIPDataTotal(voice)	Number of total voice data sent to IP side
SendToLocalDataTotal(voice)	Number of total voice data sent to local side
FSSend_ReceiveDataTotal(fax)	Number of total fax data fast received or sent
DiscardDataTotal(fax)	Number of total fax data dropped
SendToIPDataTotal(fax)	Number of total fax data sent to IP side
SendToLocalDataTotal(fax)	Number of total fax data sent to local side

1.1.44 display voice vpp

Syntax

display voice vpp [ channel channel-number ]

View

Any view

Parameter

channel-number: The voice channel number.

Description

Use the display voice vpp command to view all the statistic information in the VPP module.

This command is used to display the number of times of correct and incorrect connection, the number of times of correct and incorrect disconnection, the number of times of correct and incorrect code data receiving, the number of times of correct and incorrect code data transmission, the number of times of correct and incorrect IPP data receiving, the number of times of correct and incorrect IPP data transmission, the total number of bytes of code data received, and the total number of bytes of IPP data received in various voice channels.

Example

# Display all the statistic information of channel 0 in the VPP module.

[VG] display voice vpp channel 0

Channel = 0 Status = CH_IDLE

ConnectRightTimes = 0 ConnectWrongTimes = 0

DisConnectRightTimes = 0 DisConnectWrongTimes = 0

RecvCodecmDataRightTimes = 0 RecvCodecmDataWrongTimes = 0

SendCodecmDataRightTimes = 0

RecvIppDataRightTimes = 0 RecvIppDataWrongTimes = 0

SendIppDataRightTimes = 0 SendIppDataWrongTimes = 0

RecvCodecmDataBytes = 0 RecvIppDataBytes = 0

TimeJitterLess10msTimes = 0 TimeJitterLess20msTimes = 0

TimeJitterLess30msTimes = 0 TimeJitterLess40msTimes = 0

TimeJitterLess50msTimes = 0 TimeJitterLess60msTimes = 0

TimeJitterLess70msTimes = 0 TimeJitterLess80msTimes = 0

TimeJitterLess90msTimes = 0 TimeJitterLess100msTimes = 0

TimeJitterLess110msTimes = 0

RecvIppDataSeqHopeTimes = 0 RecvIppDataDisorderTimes = 0

RecvIppDataRecvSeqLessTimes = 0 RecvIppDataSeqMoreTimes = 0

ulSendNoBDTimes = 0 ulRecvExpirePacketTimes = 0

ulRecvDuplicatePacketTimes = 0 ulJitterBufferOverFlowTimes1 = 0

ulJitterBufferOverFlowTimes2= 0 ulEmptyPacketTimes = 0

ulSendDSPVOIPPacket = 0 ulSendDSPFOIPPacket = 0

ulCorputTimes = 0

Table 1-18 Description on the fields of the display voice vpp command

Item	Description
ConnectRightTimes	Times of connecting correctly
ConnectWrongTimes	Times of connecting wrongly
DisConnectRightTimes	Times of disconnecting correctly
DisConnectWrongTimes	Times of disconnecting wrongly
RecvCodecmDataRightTimes	Times of receiving compression/decompression data correctly
RecvCodecmDataWrongTimes	Times of receiving compression/decompression data wrongly
SendCodecmDataRightTimes	Times of sending compression/decompression data correctly
RecvIppDataRightTimes	Times of receiving IPP data correctly
RecvIppDataWrongTimes	Times of receiving IPP data wrongly
SendIppDataRightTimes	Times of sending IPP data correctly
SendIppDataWrongTimes	Times of sending IPP data wrongly
RecvCodecmDataBytes	Number of bytes of receiving compression/decompression data
RecvIppDataBytes	Number of bytes of receiving IPP data
TimeJitterLess(10-110)msTimes	Statistics of Jitter time of receiving IPP data
RecvIppDataSeqHopeTimes	Times of receiving IPP data whose sequence is accordant with expectation
RecvIppDataDisorderTimes	Times of receiving IPP data which is out of order
RecvIppDataRecvSeqLessTimes	Times of receiving IPP data whose sequence is less than expectation
RecvIppDataSeqMoreTimes	Times of receiving IPP data whose sequence is larger than expectation
ulSendNoBDTimes	Times of sending data without buffer
ulRecvExpirePacketTimes	Times of receiving expired packets
ulRecvDuplicatePacketTimes	Times of receiving duplicate packets
ulJitterBufferOverFlowTimes1	Times 1 of JitterBuffer overflows
ulJitterBufferOverFlowTimes2	Times 2 of JitterBuffer overflows
ulEmptyPacketTimes	Times of empty packets
ulSendDSPVOIPPacket	Number of voice packets sending to DSP
ulSendDSPFOIPPacket	Number of fax packets sending to DSP
ulCorputTimes	Times of collision in JitterBuffer

1.1.45 dot-match

Syntax

dot-match { end-only | left-right | right-left }

undo dot-match

View

Voice number-substitute view

Parameter

end-only: Reserves the digits that correspond to all the dots “.” at the end of the number input format.

left-right: Reserves from left to right the digits that correspond to the dots in the number input format.

right-left: Reserves from right to left the digits that correspond to the dots in the number input format.

Description

Use the dot-match command to configure dot match rules of the number-substitute rules.

Use the undo dot-match command to restore the default value. The configuration of this command only applies to the rules of the number-substitute list in the current view.

By default, the dot match rule is set to end-only.

According to the configuration of the dot-match command, the dots can be reserved by quantity and position of the dots configured in output format of the number-substitute rules, There are three dot match modes in the number-substitute rules.

l Reserve the digits that correspond to all the dots at the end of the number input format. It is the default mode.

l Reserve from left to right the digits that correspond to the dots in the number input format.

l Reserve from right to left the digits that correspond to the dots in the number input format.

The dots “.” here are virtual match digits. Virtual match digits are the digits that match the variable part (such as + % ! [ ]) in an expression. For example, when 1255 is matched with 1[234]55, the virtual match digit is 2; when it is matched with 125+, the virtual match digits are 55; when it is matched with 1..5, the virtual match digits are 25.

& Note:

For the detailed description about the “.” symbol in the number-substitute rule, refer to the rule command.

Example

# Set the dot match rule of the number-substitute list 20 to right-left.

[VG-voice-dial] number-substitute 20

[VG-voice-dial-substitute20] dot-match right-left

1.1.46 dtmf gain

Syntax

dtmf gain { hf | lf } value

und dtmf gain { hf | lf }

View

Voice subscriber-line line

Parameter

hf: Specifies the high frequency signal gain from -6.0 dB to 14.0 dB and allows one decimal digit of precision. The default value is 0.

lf: Specifies the low frequency signal gain from -6.0 dB to 14.0 dB and allows one decimal digit of precision. The default value is 0.

Description

Use the dtmf gain command to set the DTMF high and low frequency signal gains.

Use the undo dtmf gain command to restore the default value.

DTMF comprises high and low frequency signals. Set the signal gains to adjust the range of the transmit DTMF signals based on which PBX detects the telephone number. Generally you need not to adjust these gains. But the increase of the transmit gain on FXO interface may result in the DTMF overflow, so you should lessen the DTMF signal gain. The gains of high and low frequency signals both range from -6.0 dB to 14.0 dB. As the level offset cannot be so tremendous, it is recommended that you use these two commands together and specify the same gain value.

Example

# Lessen the DTMF high and low frequency signal gains.

[VG-voice-line0] dtmf gain hf -1

[VG-voice-line0] dtmf gain lf -1

1.1.47 dtmf sensitivity-level

Syntax

dtmf sensitivity-level { high | low }

undo dtmf sensitivity-level

View

Voice subscriber-line view

Parameter

high: Specifies high sensitivity level of DTMF code detection. This mode gives low reliability and may lead to erroneous DTMF code detection.

low: Specifies low level of DTMF code detection sensitivity. This mode gives high reliability but may lead to overlooked DTMF code detection.

Description

Use the dtmf sensitivity-level command to configure the sensitivity level of DTMF code detection.

Use the undo dtmf sensitivity-level command to restore the default setting of the sensitivity level of DTMF detection.

By default, the DTMF code detection sensitivity level is “high”.

Example

# Set the DTMF code detection sensitivity for voice subscriber-line line0 to low level.

[VG-voice-line0] dtmf sensitivity-level low

1.1.48 dtmf threshold (Analog Voice Subscriber-line)

Syntax

dtmf threshold index value

undo dtmf threshold index

View

Voice subscriber-line view

Parameter

index: Index corresponding to threshold, being an integer from 1 to 12.

value: Threshold value of an index.

The threshold of index is the description of the parameters:

l 0: Lower limit of the summation of ROWMAX and COLMAX. The input signals must meet the condition (ROWMAX+COLMAX) > for 0 to be identified as DTMF number, otherwise, it is regarded as low signal strength. 0 can be an integer between 0 and 5000 and the default value is 1400. The sensitivity decreases while the reliability increases as the value becomes greater.

l 1: Upper limits of maximum of ROWMAX or COLMAX used to detect the pause between two DTMF numbers. When detecting a DTMF number, it is regarded to stop only when it meets the condition max (ROWMAX, COLMAX) < 1. 1 can be an integer between 0 and 5000 and the default value is 458. The sensitivity decreases while the reliability increases as the value becomes greater.

l 2: Lower limit for a ratio of COLMAX to ROWMAX when COLMAX < ROWMAX. For an ideal DTMF number, there cannot be many differences between COLMAX and ROWMAX. The input number is mistaken for a DTMF number when it meets the condition (COLMAX / ROWMAX) > 2. It can be an integer between -18 and -3 in dB and the default value is -9. The sensitivity decreases while the reliability increases as the value becomes greater.

l 3: Lower limit for a ratio of ROWMAX to COLMAX when COLMAX >= ROWMAX. It functions similarly to the above parameter except the ratio order. The value of the input signal must be greater than this limit for the signal to be identified as a DTMF number. 3 can be an integer between -18 and -3 in dB and the default value is -9. The sensitivity decreases while the reliability increases as the value becomes greater.

l 4: Upper limit for a ratio of the second great energy in row frequency group to ROWMAX. The value of the input signal must be smaller than this limit for the signal to be identified as a DTMF number. 4 can be an integer between -18 and -3 in dB and the default value is -9. The sensitivity decreases while the reliability increases as the value becomes greater.

l 5: Upper limit for a ratio of the second energy in column frequency group to COLMAX. The value of the input signal must be smaller than this limit for signal to be identified as a DTMF number. 5 can be an integer between -18 and -3 in dB and the default value is -9. The sensitivity decreases while the reliability increases as the value becomes greater.

l 6: Upper limit for a ratio of ROW2nd to ROWMAX. The input signal must be meet the condition (ROW2nd / ROWMAX) < 6 if the signal is identified as a DTMF number. 6 can be an integer between -18 and -3 in dB and the default value is -3. The sensitivity decreases while the reliability increases as the value becomes greater.

l 7: Upper limit for a ratio of COL2nd to COLMAX. The ratio must be smaller than this threshold for the input signal to be identified as a DTMF number. 7 can be an integer between -18 and -3 in dB and the default value is -12. The sensitivity decreases while the reliability increases as the value becomes greater.

l 8: Upper limit for a ratio of a greater energy of two extra specified frequency points to max (ROWMAX, COLMAX). The ratio must be smaller than this threshold for the input signal to be identified a DTMF number. 8 can be an integer between -18 and -3 in dB and the default value is -12. The sensitivity decreases while the reliability increases as the value becomes greater.

l 9: Lower limit for a DTMF signal lasting time. The time that the DTMF key tone lasts must be larger than this threshold for the input signal to be identified as a DTMF number. 9 can be an integer between 30 and 150 in millisecond and the default value is 30. The sensitivity decreases while the reliability increases as the value becomes greater.

l 10: Frequency of a first extra specified detection frequency point. It can be an integer between 300 and 3400 in Hz and the default value is 300. This frequency should not be included in the column and row frequency group.

l 11: Frequency of a second extra specified detection frequency point. It can be an integer between 300 and 3400 in Hz and the default value is 3200. This frequency should not be included in the column and row frequency group.

l 12: Lower limit of an input signal range. Its average range must be greater than this threshold for the input signal to be identified as a DTMF number. It can be an integer between 0 and 700 and the default value is 375. The sensitivity decreases while the reliability increases as the value becomes greater. This threshold specifies a time field to exclude the situation that the noise with small signal range is detected by error.

Description

Use the dtmf threshold command to configure the sensitivity of DTMF detection.

Use the undo dtmf threshold command to restore the default settings.

The dtmf threshold command issues the thresholds for DTMF dial tone detection to the underlying layer DSP for the purpose of tuning detection sensitivity and reliability of the device. Inside the DSP, a set of generic default values have been configured. They are 1400, 458, -9, -9, -9, -9, -3, -12, -12, 30, 300, 3200, 375, with their index being 0 through 12. Professionals can use this command to adjust the device when DTMF digit detection fails. In normal cases, the defaults are adequate.

When the value2 argument is set to 0 or DTMF digit detection to insensitivity, the neglect probability is decreasing and the detection error probability is increasing as the DTMF digit collection tolerance becomes larger.

DTMF digit detection is implemented by calculating the spectrum of the input voice signal. Its spectrum shape is restricted to the configured thresholds. A DTMF dial tone is regarded valid only when all the constraints are met. To understand this, you must be aware of DTMF dial tone, as shown in Figure 1-1:

Figure 1-1 Spectrum of keys

The tone of each telephone digit is composed of two single-frequency tones. For example, the tone of the digit 1 is compounded by two sine wave signal tones of 697 Hz and 1209 Hz. A valid key tone must last at least 45 milliseconds and have an inter-digit delay of 23 milliseconds. Refer to ITU-T Recommendation Q.24 for full information.

Figure 1-2 Spectrum of the key 1

Figure 1-2 illustrates the spectrum of key 1. Compared with other frequency points, the energy levels at frequency points of 697 Hz and 1209 Hz are relatively greater. The underlying layer DSP regards { 1209 Hz, 1336 Hz, 1477 Hz, 1633 Hz } as column frequency group and { 697 Hz, 770 Hz, 852 Hz, 941Hz } as row frequency group. Each DTMF key tone is composed of one column frequency and one row frequency. The DSP module determines whether the input voice signal is a valid DTMF digit by its energy at the above eight frequencies and their double frequencies. The maximum energy in the row frequency group is ROWMAX, its double energy is ROW2nd; the maximum energy in the column frequency group is COLMAX and its double energy is COL2nd.

Example

# Set DTMF threshold 9 in voice subscriber-line view.

[VG-voice-line0] dtmf threshold 9 30

# Restore the default of DTMF threshold 9 in voice subscriber-line view.

[VG-voice-line0] undo dtmf threshold 9

1.1.49 echo-canceller

Syntax

echo-canceller { enable | tail-length milliseconds | parameter { convergence-rate value | max-amplitude value | mix-proportion-ratio value | talk-threshold value } }

undo echo-canceller { enable | tail-length }

View

Voice subscriber-line view

Parameter

enable: Enables the echo-canceller function. By default, the echo canceller function is enabled.

tail-length milliseconds: Configures the echo duration, namely the interval between the time the user utters a sound and the time when the user hers the echo. The value range of milliseconds is 0 to 64, in millisecond (ms), and the default value is 32 ms.

convergence-rate value: Sets the rate at which the amplitude of comfort noise rises. The value range is 0 to 10, and the default value is 0. The bigger the value is, the faster noise convergence will be.

Increase this value if echoes or the background noise of the opposite side are heard when only person talks. However, if this variable is too big, the noise will be less smooth.

max-amplitude value: Sets the maximum amplitude of comfort noise. The value range is 0 to 2048, and the default value is 256. The bigger this value is, the higher the noise amplitude will be. When it is set to 0, the system will implement non-linear processing only, without applying comfort noise.

Increase this value when the environment noise is high. However, if this variable is too big, the noise will be less smooth.

mix-proportion-ratio value: Sets the mixing proportion ratio control factor for comfort noise. The value range is 0 to 3000, and the default value is 100. The bigger this value is, the bigger the proportion noise will be in noise-voice mixing.

Increase this value if echoes are heard when only person talks. However, if this variable is too big, “intermittent” voice may occur.

talk-threshold value: Sets the two-way talk judgment threshold. The value range is 0 to 2, and the default value is 1.

Increase this value if echoes appear when both parties talk at the same time. However, if this value is too big, the filter coefficient convergence will become slow.

Description

Use the echo-canceller command to configure echo-canceller function parameters.

Use the undo echo-canceller command to cancel the echo-canceller function parameters setting,

During a telephone conversation, echoes appear when the voice of the users is repeated due to the line’s reasons and transmitted back to the earpiece. The reason is that the analog voice signals have leaked into the users’ own receiving path. The echo canceller function provided by the VGs can help, to a certain extent, resolve this problem.

If too large a time value is set, the converge time of the echo canceling on the network will become longer, so when the connection has just been established, the user may hear the echo; and if the time value is set as too small, the user may also hear part of the echo, because the relatively longer echo has not been completely cancelled.

Signal leak takes place only in the analog circuit part in the voice calling path, and not on a digital network.

& Note:

l The echo-canceller enable command is effective only if it is used with the echo-canceller tail-length command.

l This command is applicable to FXO and FXS interfaces.

Related command: subscriber-line.

Example

# Set the echo-canceller sampling time as 24 ms on subscriber line 1.

[VG-voice-line1] echo-canceller enable

[VG-voice-line1] echo-canceller tail-length 24

1.1.50 entity

Syntax

entity number { voip | pots }

undo entity { number | all | voip | pots }

View

Voice dial program view

Parameter

number: Identify a voice entity, The value range is from 1 to 2147483647.

all: All dial program entities

voip: Key character, indicating that this voice entity is on the IP network side for mapping the voice devices on the network side.

pots: Key character, indicating that this voice entity is a local one for mapping the local voice devices.

Description

Use the entity command to configure a voice entity and enter its view (at the same time specify the working mode related to voice).

Use the undo entity command to cancel an existing voice entity.

In a global view, use the entity command to enter a Voice entity view, and use quit to return to the dial program view.

& Note:

When configuring VoIP and POTS voice entity, different numbers should be used to identify.

Example

# Create and enter the Voice entity view to configure a POTS voice entity whose identification is 10.

[VG-voice] dial-program

[VG-voice-dial] entity 10 pots

1.1.51 fast-connect

Syntax

fast-connect

undo fast-connect

View

VoIP voice entity view

Parameter

None

Description

Use the fast-connect command to enable the fast call connection.

Use the undo fast-connect command to disable the fast connection.

By default, the fast connection is disabled.

As there is no ability negotiation for fast connection, the ability confirmation of the two parties is determined by the called gateway. When the VG acts as a calling gateway, one can set whether or not to apply fast connection mode for each originated call. If the calling gateway adopts fast connection, the called gateway will adopt it, too. Otherwise, neither one will do so.

Fast connection procedure will be used when the calling and called parties both support the function of fast connection. Provided that neither the calling nor the called gateway supports fast connection mode, the system will automatically switch to non-fast connection procedure to resume the call.

It is OK to only configure fast-connect command for VoIP voice entity on the calling gateway. Just after successfully enabling the fast connection can the tunnel function be configured.

Related command: outband, tunnel-on and voip call-start.

Example

# Enable fast connection for VoIP voice entity 10.

[VG-voice-dial-entity10] fast-connect

1.1.52 first-rule

Syntax

first-rule rule-number

undo first-rule

View

Voice number-substitute view

Parameter

rule-number: Serial numbers of the number-substitute rules, ranging from 0 to 127.

Description

Use the first-rule command to configure the number-substitute rule that is first used in the current number-substitute list.

Use the undo first-rule command to restore the default number-substitute rule.

By default, the configured rule with the smallest serial number is used first.

In a voice call, the system first uses the rule that is defined by the first-rule command when it begins to use the number-substitute rules. If this rule fails, it will try all other rules in order, till it finds the one that works or till it confirms all the rules do not work.

Example

# Set rule 4 as the first-used rule in number-substitute list 20.

[VG-voice-dial] number-substitute 20

[VG-voice-dial-substitute20] rule 4 663 3

[VG-voice-dial-substitute20] first-rule 4

1.1.53 fxo-monitoring

Syntax

fxo-monitoring enable

undo fxo-monitoring enable

View

Voice view

Parameter

None

Description

Use the fxo-monitoring enable command to enable online detection on all FXO ports of the device.

Use the undo fxo-monitoring enable command to disable online detection on all FXO ports of the device.

Online detection works only on the FXO port but not FXS ports. For VG 10-41 (version C), online detection monitors FXO online status in realtime. For VG 21-08 (version C), online detection monitors FXO online status only when calls are proceeding.

By default, online detection is enabled on a device.

Example

# Disable online detection on all FXO ports of the device.

[VG-voice] undo fxo-monitoring enable

1.1.54 hookoff-mode

Syntax

hookoff-mode { immediate | delay }

undo hookoff-mode

View

FXO Voice subscriber-line view

Parameter

immediate: Configure the immediate hookoff mode to be applied to FXO voice subscriber-line.

delay: Configure the delayed hookoff mode to be applied to FXO voice subscriber-line.

Description

Use the hookoff-mode command to configure the hookoff mode to be applied to FXO voice subscriber-line.

Use the undo hookoff-mode command to restore the default.

By default, the immediate hookoff mode is applied to FXO voice subscriber-line.

If the immediate mode is applied, the FXO interface changes immediately into offhook status after receiving the call requirement. Then user dials a second number.

If the delay mode is applied, user needs to configure dedicated line number. The system automatically calls the called according to the dedicated line number configured for the FXO interface. After the called phone is hooked off, the communication functions over the FXO voice subscriber-line.

& Note:

In delay mode, if the dedicated line number is not configured, the call setup process is the same as in immediate mode.

This command is only available to FXO voice subscriber-line users.

Example

# Configure the delayed hookoff mode to be available to FXO voice subscriber-line.

[VG-voice-line4] hookoff-mode delay

1.1.55 impedance

Syntax

impedance impedance-mode

undo impedance

View

Voice subscriber-line view

Parameter

impedance-mode: Different impedance mode corresponds to different impedance value. The following table lists all impedance modes available.

Table 1-19 Impedance modes list

Impedance mode	Matching country standard	Impedance mode	Matching country standard
Australia	Australia	Netherlands	Netherlands
Austria	Austria	Norway	Norway
Belgium-Long	Belgium	Portugal	Portugal
Belgium-Short	Belgium	R550	-
Brazil	Brazil	R600	-
China	China	R650	-
Czech-Republic	Czech	R700	-
Denmark	Denmark	R750	-
ETSI-Harmonized	ETSI-Harmonized	R800	-
Finland	Finland	R850	-
France	France	R900	-
German-Swiss	German-Swiss	R950	-
Greece	Greece	Slovakia	Slovakia
Hungary	Hungary	Spain	Spain
India	India	Sweden	Sweden
Italy	Italy	U.K.	U.K.
Japan	Japan	US-Loaded-Line	USA
Korea	Korea	US-Non-Loaded	USA
Mexico	Mexico	US-Special-Service	USA

Description

Use the impedance command to configure impedance mode for voice users.

Use the undo impedance command to restore the default impedance mode.

By default, impedance mode China is specified for the voice subscriber line.

Different subscriber line interfaces match different impedance values, and almost every country has their own standard for impedance value. So you need to choose right impedance mode depending on the specific requirements.

& Note:

This command is available to VG 10-40, G 10-41 and VG 21-08 only.

Example

# Configure the impedance mode for subscriber line 4 to Sweden.

[VG] subscribe-line 4

FXO interface encountered

[VG-voice-line4] impedance Sweden

1.1.56 ip-precedence

Syntax

ip-precedence tos-value

undo ip-precedence

View

Voice entity view

Parameter

tos-value: Indicate the IP data packet precedence level, whose value is the integer between 0 and 7. The default tos-value is 0.

Description

Use the ip-precedence command to specify the precedence level (ToS field) of the voice and fax IP packets related to a voice entity.

Use the undo ip-precedence command to restore the default value of the voice and fax IP packets related to a voice entity.

In the command, when the value of tos-value is between 1 and 5, the configuration is set for the IP data flow, and values 6 and 7 mean updating network and backbone routings. For instance, in order to ensure the voice traffic related to voice entity 103 which is of the VolP type, the IP data packet precedence level is defined as 5. Thus, when an IP call is related to this voice entity 103, the TOS field of the IP header of all the data packets transmitted to the IP network through this voice entity is set to 5. When the network receives these packets that have been set with an accepted precedence level, these packets will be processed before those packets with relatively lower precedence levels.

Related command: entity.

Example

# Configure the precedence level of the voice IP data packets to 5 for the VoIP voice entity 103.

[VG-voice-dial-entity103] ip-precedence 5

1.1.57 line

Syntax

line line-number

undo line

View

POTS voice entity view

Parameter

line-number: The number of a subscriber line.

Description

Use the line command to configure associating the voice entity with a specified voice subscriber-line.

Use the undo line command to cancel this association.

This command can only be used in POTS Voice entity view. It is effective only to the analog interfaces (FXS and FXO) and digital interfaces such as R2, PRI and digital E&M.

Figure 1-3 Count of voice subscriber-line number of voice card

The FXS and FXO ports on VG are uniformly numbered. As shown above, the number of the FXO subscriber-line next to FXS subscriber-line is 4.

Example

# Configure the voice subscriber line 1 to be corresponding to voice entity 10.

[VG-voice-dial-entity10] line 1

1.1.58 match-template

Syntax

match-template match-string

undo match-template

View

Voice entity view

Parameter

match-string: Match template. Its format is [ + ] { string [ T ] [ $ ] | T }, with the largest length of 31 characters. The characters are described in the following.

l +: Appears at the beginning of a calling number to indicate that the number is E.164-compliant.

l $: Is the last character, indicating the end of the number. That means the entire called number must match all the characters before “$” in the string.

l T: Timer. It means the system is waiting the subscriber for dialing any number till: the number length threshold is exceeded; the subscriber inputs the terminator; or the timer expires. It seems to subscribers that T matches any number in any length.

l string: A string composed of any characters of “0123456789#*.!+%[]()-”. The meanings of the characters are described in the following table:

Table 1-20 Meanings of the characters in string

Character	Meaning
0-9	Each digit, among 0 and 9, represents a digit.
# and *	Each represents a valid digit.
.	Wildcard, which can match any digit of a valid number. For example, “555. . . .” matches any number string that starts with 555 and has four additional characters.
-	Connector, used to connect two values (it is preceded by the smaller one and followed by the larger one) to express a range. For example, “1-9” represents the range of 1 to 9 (including 1 and 9).
[ ]	Represents a character selection range. It can be used together with “!” “%”, or “+”. For example, “[235-9]” represents only one single character of “2” or “3”, or between “5” and “9” can be matched.
( )	Represents a group of characters. For example, “(086)” represents the character string “086”. This pair is usually used together with “!” “%” or “+”. For example, “(086)!010” can match the two character strings “086010” and “010”.
!	Specifies that the preceding sub-expression can be absent or present once. For example, “(010)!12345678” can match “12345678” and “01012345678”.
+	Specifies that the preceding sub-expression can be present one or more times. However, when it appears at the beginning of a whole number, it means that this number is an E.164 number. In this case, the “+” character neither represents any specific number nor means number repetition. For example, “(1)9876(54)+” can match “987654”, “98765454”, “9876545454” and so on. “(2)+110022” means that 110022 is an E.164 number.
%	Specifies that the preceding sub-expression can be absent or present multiple times. For example, “9876(54)%” can match “9876”, “987654”, “98765454”, “9876545454” and so on.

& Note:

l The character or characters in front of "!”, “%”, and “+” are not to be matched accurately. They are handled similar to the wildcard “.”. Moreover, these symbols cannot be used alone. There must be a valid digit or digits in front of them.

l The control characters “!” “+” and “%” mean that the preceding sub-expression (a character or a group of characters) can be present for specific times. For example, “(100)+” can match 100, 100100, 100100 and so on. During the number matching, a number is considered completely matched if it matches any one of the numbers. In the case of longest matching, the system does not wait for succeeding dialed numbers when the number is completely matched. Refer to the T-mode related sections for circumstances where the system needs to wait for succeeding dialed numbers.

l If you want to use “[ ]” and “( )” at the same time, you must use them in the format “( [ ] )”. Other formats, such as “[ [ ] ]” and “[ ( ) ]” are illegal.

l “-“ can only be used in “[ ]”, and it only connects the same type of characters, such as “0-9”. The formats like “0-A” are illegal.

Description

Use the match-template command to configure the match template for a voice entity.

Use the undo match-template command to delete this configuration.

The match template defined by the match-template command can be used to match the number reaching the corresponding voice entity. The voice entity will complete the call if the match is successful. The match template can be defined flexibly. It can not only be a string of a unique number like 01016781234, but also an expression that can match a group of numbers, such as 010[1-5]678…. They are used to match the actual numbers in the received call packets to complete the calls.

& Note:

Comware software does not check the validity of E.164 numbers.

Example

# Set 5557922 as the telephone number of voice entity 10.

[VG-voice-dial-entity10] match-template +5557922

# Set 66.... as the match template of voice entity 20.

[VG-voice-dial-entity20] match-template 66....

# Set the match template for the numbers beginning with 661, 662, 663, and 669, and containing four other digits.

[VG-voice-dial-entity1] match-template 66 [1-3-9]....

# Set the match template for the numbers beginning with 66 and 6602 and containing four other digits.

[VG-voice-dial-entity1] match-template 66(02)!....

# Set the match template for the numbers beginning with 66 and within the length of 31 digits.

[VG-voice-dial-entity1] match-template 66T

# Set the match template for the numbers beginning with any number and within the length of 31 digits.

[VG-voice-dial-entity1] match-template T

1.1.59 max-call (voice dial program)

Syntax

max-call group max-number

undo max-call { all | group }

View

Voice dial program view

Parameter

group: Specifies a max-call group. Its value ranges from 1 to 2147483647. You can configure up to 128 groups.

max-number: Specifies the maximum number of call connections for a max-call group. It ranges from 1 to 120.

all: All max-call groups.

Description

Use the max-call command to configure max-call groups (128 groups at most).

Use the undo max-call command to delete the specified group or all the max-call groups.

This command is used to limit the number of call connections for a voice entity or a group of voice entities. It must be used with the max-call command in voice entity view: It defines the serial number of a max-call group and the maximum call connections; while the max-call command in voice entity view binds that voice entity to the max-call group with the serial number.

Related command: max-call (voice entity).

Example

# Set the maximum number of call connections of a max-call group 1 to 5.

[VG-voice-dial] max-call 1 5

1.1.60 max-call (voice entity)

Syntax

max-call group

undo max-call

View

Voice entity view

Parameter

group: Specifies a max-call group. Its value ranges from 1 to 2147483647.

Description

Use the max-call command to bind a max-call group specified by group.

Use the undo max-call command to delete the binding. Each voice entity can only be bound to one max-call group, but you can change the binding.

By default, no max-call group is bound. That means there is no limitation on the number of call connections.

This command is used together with the max-call command in voice dial program view: the max-call command in voice dial program view sets the serial number of max-call group and the maximum number of call connections; this command binds the voice entity to the max-call group with the serial number.

Related command: max-call (voice dial program).

Example

# Bind voice entity 10 to max-call group 1.

[VG-voice-dial-entity10] max-call 1

1.1.61 number-match

Syntax

number-match { longest | shortest }

View

Voice dial program view

Parameter

longest: Indicate to perform the match according to the longest number.

shortest: Indicate to perform the match according to the shortest number.

Description

Use the number-match command to configure a global number match-policy.

By default, match is performed according to the shortest number.

Command number-match is used to decide if the match is performed according to the longest or the shortest number for the number match. For instance, match-template 0106688 and match-template 01066880011 are respectively configured in two voice entities. when the user dials 01066880011, if the VG is configured with the shortest number match-policy, then the VG originates a connection to 0106688 at the remote end, and the four numbers 0011 will not be processed; if the VG is configured with the longest number match-policy, and the user only dials 0106688, the VG will wait for the user to dial. After timeout, the number match-policy configured with the system is neglected, and the shortest number match-policy will be automatically followed to make a call; if the user dials 0106688# (here the "#” represents the dial terminator configured by the system), the VG will likewise neglect the number match-policy configured by the system and use the shortest number match-policy, thus providing a greater flexibility for the configuration of user dial scheme.

Related commands: match-template, terminator.

Caution:

The configuration of the global number match policy will affect the match rules of voice entities and access numbers. For details, refer to the description of the gw-access-number command.

Example

# Configure that the number match is performed according to the longest number.

[VG-voice-dial] number-match longest

1.1.62 number-substitute

Syntax

number-substitute list-number

undo number-substitute { list-number | all }

View

Voice dial program view

Parameter

list-number: Serial number of Number-substitute list. Its value ranges from 1 to 2147483647.

all: All number-substitute lists.

Description

Use the number-substitute command to create a number-substitute list and enter voice number-substitute view.

Use the undo number-substitute command to delete the specified number-substitute list or all the number-substitute lists.

By default, no number-substitute list is created.

Related command: rule and substitute

Example

# Enter voice number-substitute view and create a number-substitute list.

[VG-voice-dial] number-substitute 1

[VG-voice-dial-substitute1]

1.1.63 outband

Syntax

outband { h245 | h225 }

undo outband

View

Voice entity view

Parameter

h245: Indicates that DTMF code to be transmitted out-of-band in the message of H.245 protocol.

H225: Indicates that DTMF code to be transmitted out-of-band in the message of H.225 protocol.

Description

Use the outband command to configure transmission of DTMF code in the outband mode.

Use the undo outband command to restore transmission of DTMF code to inband mode.

By default, the inband transmission mode is adopted.

In either fast or non-fast connection mode, DTMF code can be directly transmitted through out-of-band of H.245 and H.225 protocols.

When fast connection mode is adopted, the party for which DTMF H.245 out-of-band transmission mode is configured decides the transmission method according to whether the tunnel function is enabled during the call. If the tunnel function is enabled, DTMF code will be encapsulated in the Facility message whose h323-message-body is set to empty for transmission. If disabled, DTMF code will be encapsulated in H.245 UserInput message for transmission.

Once outband transmission of DTMF code has been configured for interoperation with industry mainstraim VGs, in order to confirm whether or not the peer gateway can accept such transmission mode of DTMF, the ability negotiation must be fulfilled after the connection is established. Currently, H3C VG series only reactively accept pass-through transmission ability negotiation of DTMF code. In no case will it actively initiate the negotiation process mentioned above.

In order to transmit DTMF code in a pass-through mode, it is demanded, during actual application, to configure this command for VoIP voice entity on the calling gateway and also for POTS voice entity for the called gateway.

Related commands: fast-connect, tunnel-on.

Example

# Configure DTMF code outband transmission in the fast connection mode for VoIP voice entity 10.

[VG-voice-dial-entity10] fast-connect

[vg-voice-dial-entity10] outband h225

1.1.64 payload-size

Syntax

payload-size { g711 | g723 | g729 } time-length

undo payload-size { g711 | g723 | g729 }

View

Voice entity view

Parameter

g711: g711 coding/decoding. It can be g711alaw or g711ulaw.

g723: g723 coding/decoding. It can be g723r53 or g723r63. Currently, the VG 20-16/32 does not support this.

g729: g729 coding/decoding. It can be g729r8 or g729a.

time-length: Time length of the DSP packets with the specified coding/decoding mode. See Table 1-21 for its value range.

Description

Use the payload-size command to set the time length for DSP to assemble a packet, that is, the time length of each voice packet.

Use the undo payload-size command to restore the default time length of voice packets.

By default, it takes 20 milliseconds for DSP to assemble a packet with g711, and 30 milliseconds with g723 or g729.

The following table lists the time length ranges and rules for DSP to assemble a packet.

Table 1-21 The Time Length Value Range of Voice Packet

G.711

G.723

G.729

VG 20-32

VG 20-16

VG 21-08

An integral multiple of 10 in the range 20 to 30

An integral multiple of 10 in the range 20 to 180

VG 10-40

VG 10-41

An integral multiple of 30 in the range 30 to 180

Related commands: default entity payload-size, compression.

Example

# Set the time length for DSP packet assembling with g711 to 30 milliseconds.

[VG-voice-dial-entity1] payload-size g711 30

1.1.65 peer-signal-port

Syntax

peer-signal-port port-number

undo peer-signal-port

View

VoIP voice entity view

Parameter

port-number: Port number of the VoIP voice entity. It is in the range 1000 to 16000 and defaults to 1720.

Description

Use the peer-signal-port command to configure the port number of the VoIP voice entity.

Use the undo peer-signal-port command to restore the default port number of the VoIP voice entity.

This command is only available in case of the VoIP entity and no dynamic routing is adopted. If the dynamic routing is configured, such as the address ras command, the port number of the peer is sent by the GK and this command is unavailable.

Related command: address.

Example

# Set the port number of VoIP voice entity 200 to 2000.

[VG-voice-dial-entity200] peer-signal-port 2000

1.1.66 plan-numbering

Syntax

plan-numbering { called | calling } { data | isdn | national | private | reserved | telex | unknown }

undo plan-numbering { called | calling }

View

Voice entity view

Parameter

called: Called number.

calling: Calling number.

data: Data numbering plan.

isdn: Numbering plan of ISDN telephone.

national: Numbering plan of national standard.

private: Private numbering plan.

reserved: Reserved extension.

telex: Subscriber telex numbering plan

unknown: Unknown numbering plan.

Description

Use the plan-numbering command to configure the calling/called numbering plan attribute of voice entities.

Use the undo plan-numbering command to restore the default numbering plan.

By default, the calling/called numbering plan of a voice entity is set to unknown.

The numbering plans comply with ITU-T Protocol Q931.

Related command: type-number.

Example

# Set the called numbering plan of voice entity 10 to national.

[VG-voice-dial-entity10] plan-numbering called national

1.1.67 plc-mode

Syntax

plc-mode { general | specific }

undo plc-mode

View

Voice subscriber-line view

Parameter

general: Uses the general frame erasure algorithm for packet loss compensation.

specific; Uses the VG-specific mode for packet loss compensation.

Description

Use the plc-mode command to configure the packet loss compensation mode.

Use the undo plc-mode command to restore the default packet loss compensation mode.

By default, the VG-specific packet loss compensation mode is used.

This command is effective for FXO and FXS interfaces only.

Example

# Configure the voice gateway so that the general packet loss compensation mode is used.

[VG-voice-line0] plc-mode general

1.1.68 polarity-reverse

Syntax

polarity-reverse

undo polarity-reverse

View

FXS and FXO voice subscriber-line view

Parameter

None

Description

Use the polarity-reverse command to enable polarity reversal on analog voice subscriber-line.

Use the undo polarity-reverse command to disable polarity reversal.

By default, polarity reversal is disabled.

& Note:

l This command is only available to analog voice subscriber-line.

l In case of FXS interfaces, this command enables/disables sending polarity reversal messages.

l In case FXO, this command enables/disables receiving polarity reversal messages.

Example

# Enable polarity reversal on FXS voice subscriber-line 0.

[VG-voice-line0] polarity-reverse

1.1.69 priority

Syntax

priority priority-order

undo priority

View

Voice entity view

Parameter

priority-order: Priority of a voice entity. Its value ranges from 0 to 10. The smaller the number is , the higher the priority is. That means 0 is the highest priority and 10 is the lowest priority.

Description

Use the priority command to configure the priority levels for voice entities.

Use the undo priority command to restore the default priority level.

By default, the priority level is set to 0.

If you have configured priority levels for voice entities and have configured priority in voice entity select rule (see select-rule), the system will first select the voice entity of the highest priority when it initiates a call. If the voice entity of the highest priority fails, it will try those of lower priority levels to initiate the call.

Example

# Set the priority level of voice entity 10 to 5.

[VG-voice-dial-entity10] priority 5

1.1.70 private-line

Syntax

private-line string

undo private-line

View

Voice subscriber-line view

Parameter

string: The E.164 telephone number of the destination end, and it may include the numbers from 0 to 9, “*”, and "#".

Description

Use the private-line command to configure private auto-ring mode for the subscriber line and the E.164 telephone number of the destination end.

Use the undo private-line command to cancel the specified connection mode.

By default, no private auto-ring mode is configured.

This command is applicable to the analog interfaces (FXO and FXS) and digital interfaces such as R2, PRI and digital E&M.

The private-line command is used to specify a connection mode for subscriber-line. The parameter string will serve as the called number of all the calls incoming to this subscriber line, i.e., after off-hook, the user need not perform any operation and the system will dial out the string as the called number automatically.

If the private-line command is not configured, when the subscriber-line enters an off-hook status, the standard session application program will generate a dial tone until enough numbers are collected and the call process is completed.

Example

# Set an automatic dialing to 5559262 after off-hook on the subscriber line 1.

[VG-voice-line1] private-line 5559262

1.1.71 private-type

Syntax

private-type { delay | quick }

undo private-type

View

Voice subscriber-line view

Parameter

delay: Delay ring mode.

quick: Quick ring mode.

Description

Use the private-type command to set the ring mode of the voice subscriber-line.

Use the undo private-type command to restore the default ring mode of the voice subscriber-line.

By default, quick ring mode applies.

With the automatic ring feature of private lines enabled, the system automatically calls with the numbers that have been configured and as such, users do not need to dial after picking up phones.

Private line automatic ringing is implemented in two modes: quick and delay. In the first mode, the system automatically calls after a user picks up the phone; but in the second mode, the system delays five seconds before it automatically calls. Within the five seconds, if the user dials a number, the system will connect the call according to the dialed number, and disables the private line automatic ringing function.

While FXO and FXS voice subscriber-lines support both quick and delay ring modes, digital E&M voice subscriber-lines support only the quick mode.

Related command: private-line.

Example

# Set the ring mode of voice subscriber-line 1 (FXS) to delay.

[VG-voice-line1] private-type delay

1.1.72 receive gain

Syntax

receive gain value

undo receive gain

View

Voice subscriber-line view

Parameter

value: Voice input gain ranging from -6.0 to 14.0 in dB with one digit after the decimal point. By default, the value is 0 dB.

Description

Use the receive gain command to configure the gain value at the voice subscriber-line input end.

Use the undo receive gain command to restore the default value.

This command is applicable to the analog interfaces (FXS and FXO) and digital interfaces such as R2, PRI and digital E&M.

When the voice signal on the line attenuates to a relatively great extent, this command can be used to appropriately enhance the voice input gain.

Related command: transmit gain and subscriber-line.

Caution:

It is recommended that you should not adjust the gain at will because this may cause a voice call failure. If the adjustment is indeed necessary, follow the professional’s instructions.

Example

# Configure the voice input gain as 3.5 dB on subscriber line 1.

[VG-voice-line1] receive gain 3.5

1.1.73 reset voice ipp

Syntax

reset voice ipp

View

Any view

Parameter

None

Description

Use the reset voice ipp command to clear IPP statistic information.

Related command: display voice ipp.

Example

# Clear IPP statistic information.

[VG] reset voice ipp

1.1.74 reset voice rcv

Syntax

reset voice rcv

View

Any view

Parameter

None

Description

Use the reset voice rcv command to clear RCV statistic information.

Related command: display voice rcv statistic.

Example

# Clear RCV statistic information.

[VG] reset voice rcv

1.1.75 reset voice voip data-statistic

Syntax

reset voice voip data-statistic

View

Any view

Parameter

None

Description

Use the reset voice voip data-statistic command to reset statistic information of voice data.

After voice data statistic is enabled via the vqa data-statistic command, all statistic items (time for successfully searching the voice table, total number of received data packets, time for searching the table in fast and common modes and voice and fax of receive and transmit channels) perform data accumulation until all counters are cleared via the reset voice voip data-statistic command. After that, new statistic begins.

Related command: vqa data-statistic, display voice voip data-statistic.

Example

# Clear statistic information of voice data.

[VG] reset voice voip data-statistic

1.1.76 reset voice vpp

Syntax

reset voice vpp [ channel channel-number ]

View

Any view

Parameter

None

Description

Use the reset voice vpp command to clear the VPP statistic information.

Related command: display voice vpp.

Example

# Clear VPP statistic information.

[VG] reset voice vpp channel 1

1.1.77 ring-generate

Syntax

ring-generate

undo ring-generate

View

Voice subscriber-line view

Parameter

None

Description

Use the ring-generate command to enable the called gateway to generate a ring-back tone for the remote endpoint at fast-connect.

Use the undo ring-generate command to restore the default setting.

By default, the called gateway transmits in a pass-through mode the ring-back tone received on its FXO interface instead of generating a ring-back tone for the remote endpoint at fast-connect.

& Note:

The FXS voice subscriber line does not support this function.

Example

# Disable the called gateway from generating the ring-back tone for the remote endpoint at fast-connect.

[VG-voice-line] undo ring-generate

1.1.78 rule

Syntax

rule rule-tag input-number output-number

undo rule rule-tag

View

Voice number-substitute view

Parameter

rule-tag: Identifies a rule. Its value ranges from 0 to 127.

input-number: Input string for number substitute. Format of the number is [ ^ ] [ + ] string [ $ ], with the maximum length of 31 characters, as listed below:

l ^: Indicates a number must be matched from the first character. When the system matches a subscriber number with the match string, the match must begin from the first character of the match string.

l +: Appears at the beginning of a calling number to indicate that the number is E.164 compliant.

l $: Specifies that the last character of the number string must be matched. Namely, the last digit of the subscriber number must match the last character of the match string.

l string: String composed of any characters of “0123456789#*.!%”. The characters are described in the following table:

Table 1-22 Meanings of the characters in string

Character	Meaning
0-9	Each digit, among 0 and 9, represents a digit.
# and *	Each represents a valid digit.
.	Wildcard, which can match any digit of a valid number. For example, “555. . . .” matches any number string that starts with 555 and has four additional characters.
!	Specifies that the preceding sub-expression can be absent or present once. For example, “9!12345678” can match “12345678” and “912345678”.
+	Specifies that the preceding sub-expression can be present one or more times. However, when it appears at the beginning of a whole number, it means that this number is an E.164 number. In this case, the “+” character neither represents any specific number nor means number repetition. For example, “(1)98765+” can match “98765”, “987655”, “9876555” and so on. “(2)+110022” means that 110022 is an E.164 number.
%	Specifies that the preceding sub-expression can be absent or present multiple times. For example, “98765%” can match “9876”, “98765”, “987655”, “9876555” and so on.

output-number: Output string for number substitute. It is composed of any characters of “0123456789#.”, with the largest length of 31 characters. The meanings of the characters are shown in the table above. Please pay attention to the following points:

& Note:

The character or characters in front of "!”, “%”, and “+” are not to be matched accurately. They are handled similar to the wildcard “.”. Moreover, these symbols cannot be used alone. There must be a valid digit or digits in front of them.

Dots in input-number and output-number are handled in three cases:

l The dots in output-number are invalid. If you have set the dot match rule to end-only using the dot-match command (i.e., only the dots at the end of the input format are handled), the dots in output-number are discarded immediately. And the digits corresponding to all the dots at the end of input-number are added to the end of output-number.

l Excessive dots in output-number are dropped. If you have configured the dot match rule as right-left or left-right by using the dot-match command, and if the dots in output-number are on higher order digits than the dots in input-number, all the numbers corresponding to the dots in input-number will be taken to substitute, from the left to the right, the dots in output-number one by one, and the dots not substituted in output-number will be ignored, namely the dots in the right part of output-number will be dropped.

l (Numbers corresponding to) the excessive dots in input-number are dropped. If you have configured the dot match rule as right-left or left-right by using the dot-match command, and if the dots in input-number are on higher order digits than, or the same order digits with, the dots in output-number, there will be two cases: (1) In case of the right-left order, based on the order of digits of the dots in output-number, the numbers on the corresponding digits will be taken, in the right-to-left order, from the numbers corresponding to the dots in input-number to substitute the dots in output-number one by one, and the dots not taken from input-number will be dropped; (2) in case of the left-right order, based on the order of digits of the dots in output-number, the numbers on the corresponding digits will be taken, in the left-to-right order, from the numbers corresponding to the dots in input-number to substitute the dots in output-number one by one, and the dots not taken from input-number will be dropped.

Description

Use the rule command to configure the number-substitute rules.

Use the undo rule command to delete the specified number-substitute rules.

By default, no number-substitute rules are configured.

When you have successfully created a number-substitute list, you need to use the command to configure the number-substitute rules in it.

Related command: substitute, number-substitute, first-rule, and dot-match.

Example

# Configure the number-substitute rules for number-substitute list 1

[VG-voice-dial-substitute1] rule 1 ^91 1

[VG-voice-dial-substitute1] rule 2 ^92 2

[VG-voice-dial-substitute1] rule 3 ^93 3

1.1.79 select-rule rule-order

Syntax

select-rule rule-order 1st-rule [ 2nd-rule [ 3rd-rule ] [ 4th-rule ] ]

undo select-rule rule-order

View

Voice dial program view

Parameter

1st-rule: Serial number of the select rule of first priority, ranging from 1 to 4. The meanings of the serial numbers are shown in the following table.

2nd-rule: Serial number of the select rule of second priority. It cannot be the same number as 1st-rule.

3rd-rule: Serial number of the select rule of third priority. It cannot be the same number as 1st-rule or 2nd-rule.

4th-rule : Serial number of the select rule of fourth priority. It cannot be the same number as 1st-rule, 2nd-rule or 3rd-rule.

Table 1-23 Meanings of the serial numbers

Serial Number	Meaning
1	Precise match
2	Priority
3	Random select
4	Longest unused

Description

Use the select-rule rule-order command to configure the priority select rule of voice entities.

Use the undo select-rule rule-order command to restore the default value.

By default, the order of voice entity priority select rules is 1->2->3. That means precise match first, voice entity priority second, and random select last.

Using the select-rule rule-order command, you can configure three rules of different priorities at most. But you cannot configure the same rules. The order of priorities is the order of select rules. If multiple priority rules are applied, the system first select voice entities according to the rule of first priority. If that rule cannot distinguish the priorities of the voice entities, the rule of second priority will be used, and so on. If all the rules cannot distinguish voice entity priorities, the voice entity with the smallest ID will be selected first.

The select rules are described in the following:

1) Precise match. From left to right, the more digits it matches, the better the precision is. The system stops using the rule once it meets a digit that cannot be matched uniquely.

2) Priority. Voice entities are divided into 11 classes, with values ranging from 0 to 10. The smaller the value is, the higher the priority is. That means 0 is the highest priority.

3) Random select. The system selects a voice entity from a group of qualified voice entities randomly.

4) Longest unused. The longer the voice entity is unused, the higher its priority is.

There will be no collision between voice entities when random select is applied. Therefore, random select can only be used as the last rule or the only rule.

Related command: select-rule search-stop and select-rule type-first.

Example

# Set the select rule of voice entities to precise match-> priority-> longest unused.

[VG-voice-dial] select-rule rule-order 1 2 4

1.1.80 select-rule search-stop

Syntax

select-rule search-stop max-number

undo select-rule search-stop

View

Voice dial program view

Parameter

max-number: The maximum number of voice entities found, ranging from 1 to 128.

Description

Use the select-rule search-stop command to configure the maximum number of voice entities found.

Use the undo select-rule search-stop command to restore the default value.

By default, the maximum number of the search for voice entities is 128.

There might be multiple voice entities that meet the call requirements during call connection. If a voice entity fails, the system can search for other qualified ones and continue the call. The select-rule search-stop command is used to define the maximum number of qualified voice entities to be found before the search stops. If there are multiple qualified voice entities, only the one of the highest priority is used to initiate a call.

Related command: select-rule rule-order and select-rule type-first.

Example

# Configure to search 5 voice entities at most.

[VG-voice-dial] select-rule search-stop 5

1.1.81 select-rule type-first

Syntax

select-rule type-first 1st-type 2nd-type

undo select-rule type-first

View

Voice dial program view

Parameter

1st-type: Serial number of the type of the first priority, ranging from 1 to 2. The meanings of the serial numbers are shown in the following table.

2nd-type: Serial number of the type of the second priority, ranging from 1 to 2. It cannot be the same number as 1st-rule.

Table 1-24 Meaning of the serial numbers

Serial Number	Meaning
1	POTS voice entity
2	VoIP voice entity

Description

Use the select-rule type-first command to configure the type-first select rules for voice entities.

Use the undo select-rule type-first command to delete the type-first select rules.

By default, voice entities are not selected according to their types.

The command is used to configure the select order for voice entities according to the voice entity types. If multiple voice entities (of different types) are qualified for a call connection, the system will select the suitable voice entity according to the type-first rules configured by the select-rule type-first command. The order of inputting the parameters determines the voice entity type priorities, which are divided into first type and second type. The system firstly selects the first type, and then the second type.

Related command: select-rule rule-order and select-rule search-stop.

Example

# Configure to select voice entities according to the type-first rules: VoIP-> POTS

[VG-voice-dial] select-rule type-first 2 1

1.1.82 select-stop

Syntax

select-stop

undo select-stop

View

Voice entity view

Parameter

None

Description

Use the select-stop command to disable the search for voice entities.

Use the undo select-stop command to re-enable the search for voice entities.

By default, the search for voice entities is enabled.

There might be multiple qualified voice entities for a call connection. If a voice entity fails, the system can search for another one that meets the requirements and continue the call. In that case, you can use this command to configure the system to stop the search when it has found the specified voice entity.

Related command: select-rule rule-order and select-rule type-first.

Example

# Configure the system to stop the search once voice entity 10 is found.

[VG-voice-dial-entity10] select-stop

1.1.83 send-number (voice entity)

Syntax

send-number { digit-number | all | truncate }

undo send-number

View

POTS voice entity view

Parameter

digit-number: Number of least significant digits that are sent, ranging from 0 to 31. It is not larger than the total number of digits of the called number.

all: All digits of the called number.

truncate: Sends the truncated called number.

Description

Use the send-number command to configure the number sending mode.

Use the undo send-number command to restore the default mode.

By default, the truncate mode is used.

This command only applies to POTS voice entities. As for the numbers sent to PSTN, this command is used to control how to send the called numbers. You can not only configure to send some digits (defined by digit-number from right to left) or all digits of the called number, but also the truncated called numbers, i.e., the numbers that match the wildcard “.” at the end.

Related command: dot-match and match-template.

Example

# Configure voice entity 10 to send the 6 least significant digits of the called number.

[VG-voice-dial-entity10] send-number 6

1.1.84 send-number (voice subscriber-line)

Syntax

send-number

undo send-number

View

FXS voice subscriber-line view

Parameter

None

Description

Use the send-number command to configure the FXS voice subscriber-line to allow the FXS to send called numbers.

Use the undo send-number command to disable the FXS to send called numbers.

By default, the FXS is disabled to send called numbers.

Normally, the FXS interface is connected to a standard telephone, so the FXS does not need to send called numbers. But sometimes, the FXS interface is connected to the FXO interface. In this case, when the FXO receives a ring from the FXS, it imitates an off-hook and sends dial tone for the second stage of dial. With the send-number feature enabled, the FXS voice subscriber-line can send called numbers to the connected FXO interface directly to free the calling party from the second stage of dial.

& Note:

l There is the situation where the sender-number feature is enabled on the FXS subscriber-line of an FXS interface that is connected to a telephone. If the telephone is called in this case, the user can hear dial tone after picking up the phone.

l The FXS voice subscriber line of VG20-16/32 does not support the local primary dialing (FXS port sends the call number to FXO port).

Example

# Enable the FXS to send called numbers on FXS voice subscriber-line 0.

[VG-voice-line0] send-number

1.1.85 send-ring

Syntax

send-ring

undo send-ring

View

VoIP voice entity view

Parameter

None

Description

Use the send-ring command to enable the local end to send ringback tone.

Use the undo send-ring command to disable the local end from sending ringback tone.

This command is available only the fast-connect function is enabled.

By default, the local end does not send ringback tone.

Example

# Enable the local end to send ringback tone.

[VG-voice-dial-entity8801] send-ring

1.1.86 shutdown (voice entity)

Syntax

shutdown

undo shutdown

View

Voice entity view

Parameter

None

Description

Use the shutdown command to configure to change the management status of specified voice entity from UP to DOWN.

Use the undo shutdown command to restore the voice entity management status from DOWN to UP.

By default, the voice entity management status is UP.

Running command shutdown will cause the voice entity unable to make calls.

Example

# Change the status of voice entity 4 to DOWN.

[VG-voice-dial-entity4] shutdown

1.1.87 shutdown (voice subscriber-line )

Syntax

shutdown

undo shutdown

View

Voice subscriber-line view

Parameter

None

Description

Use the shutdown command to configure the voice subscriber-line from UP to DOWN.

Use the undo shutdown command to restore the voice subscriber-line from DOWN to UP.

By default, the voice subscriber-line status is UP.

This command is applicable to FXO and FXS subscriber-lines.

The function of the all voice subscriber-lines specified by the command shutdown is disabled, whereas using command undo shutdown will enable all the voice subscriber-lines.

Example

# Enable voice subscriber-line 1 of the voice subscriber-line off-line.

[VG-voice-line1] shutdown

1.1.88 silence-th-span

Syntax

silence-th-span threshold time-length

undo silence-th-span

View

FXO subscriber-line view

Parameter

threshold: Silence detection threshold ranging from 3 to 100 (default value is 20). For the difference between two adjacent sampling points of voice signal from the PBX, if its absolute value is smaller than the threshold, the system regards it as silence. Generally, the signal amplitude on the links without traffic is from 2 to 5.

time-length: Duration of the silence detection. The system hooks on automatically upon the expiration of the duration. It ranges from 20 to 7200 seconds (default value is 7200 seconds, namely 2 hours).

Description

Use the silence-th-span command to set the parameter for the auto-hook-on of the silence detection.

Use the undo silence-th-span command to restore the default value.

Hook-on is used to prevent the FXO port from hanging when the busy tone detection fails. Generally you need not to use this command to adjust the busy tone detection parameters. When configuring, try to get an appropriate value to avoid the mishook-on and resource seizure of FXO.

Example

# Set the silence detection threshold to 3 and the time to 10 seconds.

[VG-voice-line0] silence-th-span 3 10

1.1.89 special-service

Syntax

special-service { local | remote }

View

Voice dial program view

Parameter

local: Enables the special-service numbers for local users. All local special-service numbers are enabled.

remote: Enables the special-service numbers for remote users. Namely users can dial special-service numbers that start with “*”, “#”, or “*#” and end with “#”, without being restricted by the terminator.

Description

Use the special-service command to enable special-service numbers for local or remote users.

Use the undo special-service command to disable special-service numbers and cancel the change to the special-service number dial tone.

By default, the special-service numbers are disabled. VG 80-20 does not support this command.

The special-service numbers include: do-not-disturb, call transfer on busy, call transfer unconditional, alarm service, and lines group access.

l Do-not-disturb

With the “do-not-disturb” service activated, the called subscriber will refuse any incoming call, regardless of whether line is idle, and the calling party will hear the busy tone.

To activate this service on a DTMF phone set connected to the voice gateway, you can press “*56#” after hooking off. To deactivate the service, press “#56#”.

l Call transfer on busy

With the call transfer on busy service activated, a new incoming call will be transferred to a specified number if the called subscriber’s line is busy.

To activate this service on a DTMF phone set connected to the voice gateway, you can press “*58*ABCD#” after hooking off. To deactivate the service, press “#58#”.

& Note:

“ABCD” represents the telephone number to which the calls will be transferred. Note that this function is available only for the phone sets connected to the FXS port on the voice gateway. In addition, you can only specify a telephone connected to the same voice gateway as the call transfer destination. Otherwise, the setting will be invalid.

To set new special service number, you must cancel the previously configured one.

l Call transfer unconditional

With the call transfer unconditional service activated, the incoming calls will be transferred to the specified number regardless of whether the called subscriber line is busy.

To activate this service on a DTMF phone set connected to the voice gateway, you can press “*57*ABCD#” after hooking off. To deactivate the service, press “#57#”

& Note:

“ABCD” represents the telephone number to which the calls will be transferred. Note that this function is only available for the telephones connected to the FXS port on the voice gateway. In addition, you can only specify a telephone connected to the same voice gateway as the call transfer destination. Otherwise, the setting will be invalid

l Alarm call service

With the alarm service activated, the telephone will ring for 45 seconds at the time specified by the subscriber and automatically disconnect the line after that. This function is only valid for 24 hours.

To activate this service on a DTMF phone set connected to the voice gateway, you can press “*55*HHMM#” after hooking off. To deactivate the service, press “#55#”

& Note:

l “HH” represents the hour, which can be any integer from 0 to 23. “MM” represents the minute, which can be any integer from 0 to 59.

l The set time is based on the system time of the VG.

l Lines group access

With this function, multiple physical telephone lines can be configured with one telephone number or the wildcard “.”. Whenever there is an incoming call request, the system will automatically select an available line to answer. Thus, the configuration is streamlined and the networking capability is improved.

If you use the command to disable the special-service numbers, all the numbers are disabled, and vice versa.

Related command: special-service switch-dialtone.

Example

# Enable local special-service numbers:

[VG-voice-dial] special-service local

# Enable remote special-service numbers:

[VG-voice-dial] special-service remote

1.1.90 special-service switch-dialtone

Syntax

special-service switch-dialtone

undo special-service switch-dialtone

View

Voice dial program view

Parameter

None

Description

Use the special-service switch-dialtone command to enable the VG to play the special dial tone to local or remote users after the special-server number function is enabled.

When the following conditions are met, the user will hear the dial tone for special service after picking up the phone:

l The special-service numbers function is enabled for local or remote users by means of the special-service command.

l The special-service number dial tone function is enabled by means of the special-service switch-dialtone command.

l The user has set a special-service number to use a special service.

& Note:

l If the special-service number function is disabled, the special-service switch-dialtone command will be unavailable.

l If you have not enabled the dial tone for the special-service number function by means of the special-service switch-dialtone command, the user will still hear the normal dial tone even if the remote supports the special-service number function.

l VG 80-20 does not support this function.

l The undo special-service switch-dialtone command can be used only if the special-service mode is set to “local”; if the special-service mode is set to “remote”, the special-service switch-dialtone cannot be used, and the switch-dialtone parameter, if configured, will be cancelled.

Example

# Change the dial tone for special-service numbers:

[VG-voice-dial] special-service switch-dialtone

# Restore the dial tone for special-service numbers to the normal dial tone.

[VG-voice-dial] undo special-service switch-dialtone

1.1.91 subscriber-line

Syntax

subscriber-line line-number

View

Any view

Parameter

line-number: Subscriber line number.

Description

Use the subscriber-line command to enter analog subscriber-line views such as FXS and FXO and digital voice subscriber-line views such as R2, PRI and digital E&M.

In the view, use subscriber-line line-number command to enter the voice subscriber-line view. When line-number is the FXS voice subscriber-line, the system will enter the FXS voice subscriber-line view. When the line-number is the FXO voice subscriber-line, the system will enter the FXO voice subscriber-line view.

Related command: entity.

Example

# Enter subscriber line 1 configuration in voice view.

[VG-voice] subscriber-line 1

[VG-voice-line1]

1.1.92 substitute

Syntax

substitute { called | calling } list-number

undo substitute { called | calling }

View

Voice subscriber-line view, voice entity view

Parameter

called: Applies the number-substitute rules to the called number.

calling: Applies the number-substitute rules to the calling number.

list-number: Serial number of the number-substitute list. Its value ranges from 1 to 2147483647.

Description

Use the substitute command to bind the calling/called number-substitute list to the voice subscriber-line or voice entity.

Use the undo substitute command to delete the calling/called number-substitute list that is bound to the voice subscriber-line or voice entity.

By default, no number-substitute list is bound to voice subscriber-line or voice entity. That means the system does not substitute number.

Firstly, you should use the number-substitute list-number command to configure the number-substitute lists in voice dial program view, and use the rule command to configure the rules in the list. Secondly, you should use the substitute command to apply the number-substitute list to the voice subscriber-line or voice entity.

& Note:

According to network requirements, you can complete number –substitute in the following two ways.

l Before voice entities are matched, you can use the substitute command in the voice subscriber-line view to substitute the calling/called numbers corresponding to the specified subscriber-line.

l After voice entities are matched and before a call is initiated, you can use the substitute command in voice entity view to substitute the specified calling/called numbers.

Related command: number-substitute and rule.

Example

# Apply called number-substitute list 6 to voice subscriber-line 3.

[VG-voice-line3] substitute called 6

1.1.93 substitute incoming-call

Syntax

substitute incoming-call { called | calling } list-number

undo substitute incoming-call { called | calling } { all | list-number }

View

Voice dial program view

Parameter

called: Applies the number-substitute rules to the called number.

calling: Applies the number-substitute rules to the calling number.

list-number: Serial number of the number-substitute list. Its value ranges from 1 to 2147483647.

all: All number-substitute rule lists.

Description

Use the substitute incoming-call command to bind the calling/called numbers of the incoming calls from the network side to the number-substitute list.

Use the undo substitute incoming-call command to cancel the binding.

By default, no number-substitute list is bound. That means the system does not substitute any number.

Firstly, you can use the number-substitute list-number command in voice dial program view to create a number-substitute list, and then use the rule command to configure the rules in the list. Secondly, you can use the substitute incoming-call command to apply the number-substitute list to the strategy of incoming calls from the network.

You should follow these rules to use the command:

1) 32 number-substitute lists can be bound at most.

2) The system searches all the rules in the number-substitute lists in order, and it stops the search once one rule works.

& Note:

According to network requirements, before voice entities are matched, you can use the substitute incoming-call command in the voice dial program view to substitute all the calling/called numbers of the incoming calls from the network side.

Related command: number-substitute, rule, and substitute outgoing-call.

Example

# Apply number-substitute list 5 to the called numbers of the incoming calls from the network side.

[VG-voice-dial] substitute incoming-call called 5

1.1.94 substitute outgoing-call

Syntax

substitute outgoing-call { called | calling } list-number

undo substitute outgoing-call { called | calling } { all | list-number }

View

Voice dial program view

Parameter

called: Applies the number-substitute rules to the called number.

calling: Applies the number-substitute rules to the calling number.

list-number: Serial number of the number-substitute list. Its value ranges from 1 to 2147483647.

all: Deletes all number-substitute rule lists.

Description

Use the substitute outgoing-call command to bind the calling/called numbers of the outgoing calls from the network side to the number-substitute list.

Use the undo substitute outgoing-call command to cancel the binding.

By default, no number-substitute list is bound. That means the system does not substitute any number.

Firstly, you can use the number-substitute list-number command in voice dial program view to create a number-substitute list, and then use the rule command to configure the rules in the list. Secondly, you can use the substitute outgoing-call command to apply the number-substitute list to the policy of outgoing calls from the network.

The follow rules apply to this command:

l 32 number-substitute lists can be bound at most.

l The system searches all the rules in the number-substitute lists in order, and it stops the search once one rule works.

& Note:

According to network requirements, before voice entities are matched, you can use the substitute outgoing-call command in the voice dial program view to substitute all the calling/called numbers of the outgoing calls from the network side.

Related command: number-substitute, rule, and substitute incoming-call.

Example

# Apply number-substitute lists 5, 6, and 8 to the called numbers of the outgoing calls from the network side.

[VG-voice-dial] substitute outgoing-call called 5

[VG-voice-dial] substitute outgoing-call called 6

[VG-voice-dial] substitute outgoing-call called 8

1.1.95 terminator

Syntax

terminator character

undo terminator

View

Voice dial program view

Parameter

character: Telephone number terminator. The valid characters are: 0-9, “#”, “*”.

Description

Use the terminator command to configure characters as the terminator of a telephone number.

Use the undo terminator command to cancel the existing setting.

By default, no terminator is configured.

When dialing, users can end a call with a terminator. The gateway will end the number-receiving once detecting the terminator. If the users continue dialing, the gateway will discard these numbers.

Related command: match-template and timer.

Example

# Configure using “#” as a terminator.

[VG-voice-dial] terminator #

1.1.96 timer dial-interval

Syntax

timer dial-interval seconds

undo timer dial-interval

View

Voice subscriber-line view

Parameter

seconds: The maximum interval between two digits, which is in the range of 1 to 300 seconds, and the default value is 4s.

Description

Use the timer dial-interval command to configure the timer that the system waits for a subscriber to dial the next digit.

Use the undo timer dial-interval command to restore the default time settings.

This command is applied to analog interfaces (FXO and FXS) and digital interfaces such as PRI and digital E&M.

This timer will restart whenever the subscriber dials a digit and will work in this way until all the digits of the number are dialed. If the timer times out before the dialing is completed, the subscriber will be prompted to hook up and the call is terminated.

Example

# Set the maximum duration waiting for the next digit on voice line 1 to 5 seconds.

[VG-voice-line1] timer dial-interval 5

1.1.97 timer first-dial

Syntax

timer first-dial seconds

undo timer first-dial

View

Voice subscriber-line view

Parameter

seconds: The maximum time waiting for the first dial, which is in the range of 1 to 300 seconds, and the default value is 10s.

Description

Use the timer first-dial command to configure the timer that the system waiting for a subscriber to dial the first digit.

Use the undo timer first-dial command to restore the default time settings.

This command is applied to FXO, FXS subscriber-lines. If the private line call is configured to delay 5 seconds, the first-dial time cannot be more than 5 seconds. If the configured time of the timer is less than 5 seconds, the system handles it as 5 seconds.

Upon the expiration of the timer, the subscriber will be prompted to hook up and the call is terminated.

Example

# Set the maximum duration waiting for the first dial on voice line 1 to 10 seconds.

[VG-voice-line1] timer first-dial 10

1.1.98 trace interval

Syntax

trace interval [ packets ]

View

Voice view

Parameter

packets: Number of voice packets, which is in the range of 1 to 10000 and defaults to 200.

Description

Use the trace interval command to control the frequency at which the information is recorded in the process voice data debugging with the debugging voice data-flow command, that is, the number of voice packets upon the pass of which a record will be made.

Example

# Record debugging information for every 300 voice packets.

[VG-voice] trace interval 300

1.1.99 transmit gain

Syntax

transmit gain value

undo transmit gain

View

Voice subscriber-line view

Parameter

value: Voice output gain ranging -6.0 to 14.0 in dB with one digit after the decimal point. By default, the value is 0 dB.

Description

Use the transmit gain command to configure the voice subscriber-line output end gain value.

Use the undo transmit gain command to restore the default value.

This command is applicable to the analog interfaces (FXS and FXO) and digital interfaces such as R2, PRI and digital E&M.

When a relatively small voice signal power is needed on the output line, this command can be used to properly increase the voice output gain value to adapt to the output line signal requirement.

Related command: receive gain and subscriber-line.

Caution:

It is recommended that you should not adjust the gain at will because this may cause a voice call failure. If the adjustment is indeed necessary, follow the professional’s instructions.

Example

# Configure the voice output gain value as -2.0dB on subscriber line 1

[VG-voice-line1] transmit gain -2.0

1.1.100 tunnel-on

Syntax

tunnel-on

undo tunnel-on

View

VoIP voice entity view

Parameter

None

Description

Use the tunnel-on command to enable tunnel function.

Use the undo tunnel-on command to disable tunnel function.

By default, the tunnel function is disabled.

Tunnel function can assist in negotiating process of such nonstandard H.245 message as transmitting DTMF code in a pass-through mode.

Only after successfully enabling the fast connection mode, can one fulfill the configuration of tunnel function. As the calling gateway, it can be decided whether or not to enable the tunnel function for each call on the VG. Being the called gateway, it shall be decided whether or not to enable the tunnel function based on the status of the calling gateway. That is, if the function is enabled on calling gateway, it will also be enabled on the called gateway. Otherwise, tunnel function is disabled on both sides. In order to transmit DTMF code in a pass-through mode via the fast connection mode, the tunnel function must be enabled. Otherwise DTMF code cannot be transmitted.

During actual configuration, it is only necessary to fulfill this command for the VoIP voice entity at the calling gateway.

Related command: fast-connect and outband.

Example

# Enable the tunnel function for VoIP voice entity 10.

[VG-voice-dial-entity10] fast-connect

[VG-voice-dial-entity10] tunnel-on

1.1.101 type-number

Syntax

type-number { called | calling } { abbreviated | international | national | network | reserved | subscriber | unknown }

undo type-number { called | calling }

View

Voice entity view

Parameter

called: Called number.

calling: Calling number.

abbreviated: Abbreviated dial type.

international: International number type.

national: National number type.

network: Special network number type.

reserved: Extension reserved.

subscriber: Subscriber number type.

unknown: Unknown number type.

Description

Use the type-number command to configure the calling/called number types of voice entities.

Use the undo type-number command to restore the default number types.

By default, calling/called number types of voice entities are set to unknown.

The number types comply with ITU-T Recommendation Q.931.

Example

# Set the called number of voice entity 10 to national type.

[VG-voice-dial-entity10] type-number called national

1.1.102 vad-on

Syntax

vad-on

undo vad-on

View

Voice entity view

Parameter

None

Description

Use the vad-on command to enable silence detection function.

Use the undo vad-on command to disable silence detection function.

By default, the silence detection function is disabled.

VAD is an abbreviation of Voice Activity Detection, generally termed as silence detection. Its basic idea is to detect and delete any silence on the basis of the difference of energy between the voice signals of people’s conversation and their silence signals, so that no signals are produced; and only when an abrupt activity tone is detected will a voice signal be generated and transmitted.

Related command: default entity vad-on.

& Note:

The G.711 voice compensation mode does not support voice activity detection. After the voice activity detection function is enabled, if the negotiated voice compensation mode is G.711, the voice activity detection function will not work.

Example

# Enable the VAD function of POTS voice entity 10.

[VG-voice-dial-entity10] vad-on

1.1.103 vi-card busy-tone-detect

Syntax

vi-card busy-tone-detect { auto index line-number [ free | time ] | custom area-number index argu f1 f2 p1 p2 p3 p4 p5 p6 p7 }

undo vi-card busy-tone-detect custom index

View

Voice view

Parameter

index: Index of busy tone frequency, ranging 0 to 3.

line-number: Subscriber line number.

free: Free the data and stop obtaining the data.

time: Detection time in seconds, ranging from 1 to 12.

area-number: Area number. This argument is reserved currently and is set to 2.

argu: Currently reserved argument.

f1: Frequency of the single-frequecy tone, in units of Hz.

f2: Currently reserved argument.

p1: Energy threshold of the single-frequency tone.

p2: Currently reserved argument.

p3: Single-frequency tone duration in ms.

p4: Allowed error for single-frequency tone in ms.

p5: Silence duration in ms.

p6: Allowed error for silence duration in ms.

p7: Allowed error for the difference between the single-frequency tone duration and the silence duration, that is, the absolute value for the difference between p3 and p4, in units of ms.

Description

Use the vi-card busy-tone-detect command to configure busy tone detection parameters on FXO subscriber-line.

The vi-card busy-tone-detect auto command is valid for FXO interface only. It is not provided for VGs without an FXO interface. Only one channel is supported on a device at one time. Four kinds of busy tone can be saved, each is identified by index.

The vi-card busy-tone-detect command is applicable in most cases. It can facilitate busy tone detection.

Use the vi-card busy-tone-detect custom command to save DSP parameters and set customized busy tone detection parameters for the FXO port.

Use the undo vi-card busy-tone-detect custom command to restore the default values for customized busy tone detection parameters for the FXO port.

When busy tone is detected on the FXO port through the vi-card busy-tone-detect auto command, the system will calculate parameter values related to busy tone detection. You can use the display current-configuration command to display parameter values set through the vi-card busy-tone-detect custom command to help manual configuration and modification of these settings.

& Note:

The cptone parameters configured through the vi-card busy-tone-detect custom command will not take effect until you execute the area custom command in voice subscriber-line view.

Related command: vi-card custom-toneparam, vi-card cptone-custom.

Example

# Enable the automatic busy tone detection on the No.3 subscriber line for the index of 0.

[VG-voice] vi-card busy-tone-detect auto 0 3

1.1.104 vi-card cptone-custom

Syntax

vi-card cptone-custom type arg0 arg1 arg2 arg3 arg4 arg5 arg6

undo vi-card cptone-custom { type | all }

View

Voice view

Parameter

type: Cptone type. Currently supported cptone types include the dial tone, special dial tone, busy tone, congestion tone, ringback tone and waiting tone.

arg0: Frequency combination mode, ranging from 0 to 2. 0 indicates the two frequencies are overlapped, 1 indicates the two frequencies are multiplied and 2 indicates the two frequencies are alternant.

arg1/arg2: Frequecy values of the two single-frequecy tones, in units of Hz. This argument is related to the value of arg0. If arg0 is set to overlapped or alternant, the value of arg1/arg2 ranges from 300 to 3400. If arg0 is set to multiplied, the value of arg1/arg2 ranges from 0 to 3400. Note that the absolute value for sum or difference of arg1 and arg2 must be within the range between 300 and 3400.

arg3: ON duration of the first ON/OFF period, in units of ms and ranging from 30 to 8191. To continuously play the tone, set this argument to 8192.

arg4: OFF duration of the first ON/OFF period, in units of ms and ranging from 30 to 8191.

arg5: ON duration of the second ON/OFF period, in units of ms and ranging from 30 to 8191.

arg6: OFF duration of the second ON/OFF period, in units of ms and ranging from 30 to 8191.

all: Specifies all types of cptone.

Table 1-25 types of cptone

Type	Description
dial-tone	Dial tone
special-dial-tone	Special dial tone
congestion-tone	Congestion tone
busy-tone	Busy tone
ringback-tone	Ring back tone
waiting-tone	Waiting tone

Description

Use the vi-card cptone-custom command to set cptone parameters.

Use the undo vi-card cptone-custom command to remove the customized cptone parameters.

& Note:

The cptone parameters customized through the cptone-custom command will not take effect until you run the cptone cs command in voice subscriber-line view.

Example

# Customize a busy tone with single-frequecy 425 Hz and the ON and OFF durations are 350 ms respectively.

[Router-voice] vi-card cptone-custom busy-tone 0 425 425 350 350 350 350

1.1.105 vi-card custom-toneparam

Syntax

vi-card custom-toneparam area-number index arg0-arg9

undo vi-card custom-toneparam index

View

Voice view

Parameter

area-number: Reserved argument and is set to 2.

index: Index of busy tone frequency, ranging from 0 to 3.

arg0: Reserved argument, ranging from 0 to 32767. The default value is 0.

arg1/arg2: Single frequency in Hz, ranging from 50 to 3500. The default value is 450.

arg3/arg4: Energy threshold of single frequency. The value range is 50 to 5000, and the default value is 400.

arg5: Busy tone (single frequency) duration in ms. The value range is 50 to 10000 and the default value is 300.

arg6: Allowed error for busy tone (single frequency) duration in ms, and the value range is 10 to 2000. The default value is 80.

rag7: Silence (low level) duration, in ms. The value range is 50 to 2000. The default value is 300.

arg8: Allowed error for silence (low level) duration, in ms. The value range is 10 to 2000, and the default value is 80.

arg9: Allowed error for the difference between the busy tone (single frequency) duration and the silence (low level) duration, in ms. The value range is 10 to 2000, and the default value is 160.

Description

Use the vi-card custom-toneparam command to save the DSP parameters and customize the busy tone detection parameters for FXO interfaces.

Use the undo vi-card custom-toneparam command to delete the customized settings.

After the system detects a busy tone on an FXO interface with the vi-card busy-tone-detect command, it can automatically calculate the parameters related to busy tone detection. You can use the display current-configuration command to view the values of the busy tone parameters configured using the vi-card custom-toneparam command. This set of parameter values can help you configure and adjust the busy tone detection parameters.

Caution:

l This command is valid for FXO interfaces, and is not provided for VGs without an FXO interface.

l You must run the area custom command in FXO subscriber-line view before the busy tone parameters you have configured using the vi-card custom-toneparam command can take effect.

l The undo vi-card custom-toneparam command does not include the area-number argument.

Related commands: area custom, vi-card busy-tone-detect.

Example

# Configure the busy tone frequency of No.0, and enable automatic definition of the busy tone detection parameter on the FXO interface.

[VG-voice] vi-card custom-toneparam 2 0 100 500 500 300 450 300 80 300 80 160

1.1.106 vi-card hook-sensitivity

Syntax

vi-card hook-sensitivity { low | middle | high }

undo vi-card hook-sensitivity

View

Voice view

Parameter

high: Sets the hookoff detection sensitivity to high level. This mode gives high reliability and will not lead to erroneous hookoff/hookon detection, though the detection time is relatively long.

middle: Sets the hookoff/hookon detection sensitivity to middle level. This mode gives general reliability.

low: Sets the hookoff/hookon detection sensitivity to low level. This mode gives low reliability and may lead to erroneous hookoff detection, though the detection time is relatively short.

Description

Use the vi-card hook-sensitivity command to configure the hookoff/hookon detection sensitivity.

Use the undo vi-card hook-sensitivity command to restore the hookoff/hookon detection sensitivity to the default setting.

By default, the hookoff/hookon detection sensitivity is set to “middle”.

& Note:

This command is applicable to VG20-16 and VG 20-32 only.

Example

# Set the hookoff/hookon detection sensitivity to high level.

[VG-voice] vi-card hook-sensitivity high

1.1.107 vi-card polarity-reverse

Syntax

vi-card polarity-reverse { all | line number }

undo vi-card polarity-reverse { all | line number }

View

Voice view

Parameter

all: Specifies all analog interfaces.

line number: Specifies the analog voice subscriber lines. number is character string made up of characters “-”, “,” and numbers from 0 to 9. The string must not begin or end with the characters “-” or “,”. For example, if the specified analog voice subscriber line numbers are 1, 4, 5, 6, 7, 8 and 10 to write as follows: 1, 4-8, 10.

Description

Use the vi-card polarity-reverse command to enable polarity reversal on analog voice subscriber lines.

Use the undo vi-card polarity-reverse command to disable polarity reversal on analog voice subscriber lines.

By default, the polarity reversal is disabled on analog voice subscriber lines.

& Note:

l This command is available to analog voice subscriber lines only.

l In case of FXS interfaces, this command enables/disables sending polarity reversal messages.

l In case FXO, this command enables/disables receiving polarity reversal messages.

Example

# Enable polarity reversal on the specified analog voice subscriber lines 0, 3, 4, and 5.

[VG-voice] vi-card polarity-reverse line 0,3-5

1.1.108 vi-card reboot

Syntax

vi-card reboot [slot-number ]

View

Voice view

Parameter

slot-number: The slot number of the voice interface card to be reset. This parameter is effective for VG 80-20 only, and is not needed for other models.

Description

Use the vi-card reboot command to reset a voice interface card.

Example

# Reset a voice interface card.

[VG-voice] vi-card reboot

WARNING: The voice interface will be reset! Continue?(Y/N) y

1.1.109 voice-setup

Syntax

voice-setup

View

System view

Parameter

None

Description

Use the voice-setup command to enter voice view.

Example

# Enter voice view.

[VG] voice-setup

[VG-voice]

1.1.110 voip calledtunnel

Syntax

voip calledtunnel { enable | disable }

View

Voice view

Parameter

enable: Enable the tunnel function on called GW.

disable: Disable the tunnel function on called GW.

Description

Use the voip calledtunnel command to configure whether to enable the tunnel function on called GW.

Use the voip calledtunnel disable command to disable the tunnel function.

By default, the tunnel function on the called GW is enabled.

To enable interoperation between an H.323-enabled device and a device that does not support the tunneling function, you need to use this command to disable the tunneling function on the called GW. If the tunneling mode has been enabled on the calling GW, the tunneling function will be enabled/disabled on the called GW based on the voip calledtunnel command.

Related command: tunnel-on.

Example

# Disable the tunnel function of the called GW.

[VG-voice] voip calledtunnel disable

1.1.111 voip call-start

Syntax

voip call-start { fast | normal }

View

Voice view

Parameter

fast: The called GW initializes calls in a fast way.

normal: The called GW initializes calls in a non-fast way.

Description

Use the voip call-start command to configure a call initialization mode for the called GW.

By default, the fast mode is used.

As the process of facilities negotiation is omitted in fast-connection, the faculties of the two parties are determined by the GW. If a VG acts as a calling GW, you can enable or disable fast-connection for each channel of initiated calls. If it acts as a called GW, it will use or not use the fast-connection mode to initialize calls depending on the parameters of the voip call-start command, in the case that the calling GW uses the fast-connection mode.

Related command: fast-connect.

Example

# Configure the called GW to initialize calls in the non-fast mode.

[VG-voice] voip call-start normal

1.1.112 voip h323-config tcs-t38

Syntax

voip h323-config tcs-t38

undo voip h323-config tcs-t38

View

Voice view

Parameter

None

Description

Use the voip h323-config tcs-t38 command to enable the voice gateway to carry the T.38 capacity description in its capacity set when it is in H.323 slow-start mode.

Use the undo voip h323-config tcs-t38 command to disable the voice gateway to carry the T.38 capacity description in its capacity set when it is in H.323 slow-start mode.

By default, T.38 description is carried.

As NetMeeting does not support T.38 capacity description parsing, you must use the undo voip h323-config tcs-t38 command in order to work with NetMeeting.

& Note:

As this command is globally effective, the configuration makes all the voice entities carry the T.38 capacity description in their capacity sets. If interoperability with NetMeeting is required only by a voice entity, you can disable fax using the fax baudrate disable command or set the fax mode to a non-T.38 mode (pcm, for example).

Example

# Disable the voice gateway to carry T.38 capacity description in its capacity set when it is in H.323 slow-start mode.

[VG-voice] undo voip h323-config tcs-t38

1.1.113 voip h323-descriptor

Syntax

voip h323-descriptor descriptor

undo voip h323-descriptor

View

Voice view

Parameter

descriptor: The description character string of H.323, in the length of 1 to 63. By default, the descriptor is "Voice-Gateway".

Description

Use the voip h323-descriptor command to configure the H.323 description character string of the voice gateway.

It is recommended to configure with default value, also you can configure according to the actual requirements. If both ends are H3C devices, and the attribute is configured at both ends (i.e., the default value is not adopted), the character strings of both ends must be the same.

Example

# Configure the descriptor of H.323 as “mystring”.

1.1.114 voip h323-timer

Syntax

voip h323-timer { socket-create | twaitalerting | twaitconnect | twaitsetup } seconds

undo voip h323-timer { socket-create | twaitalerting | twaitconnect | twaitsetup }

View

Voice view

Parameter

socket-create seconds: Configure the duration of the socket timer, ranging from 3 seconds to 30 seconds. The default value is 30 seconds.

twaitalerting seconds: Configure the interval for waiting for the Alerting message after the calling party sends the Setup message, ranging from 3 seconds to 30 seconds. The default value is 20 seconds.

twaitconnect seconds: Configure the interval that the calling party waits for the Connect message, ranging from 3 seconds to 300 seconds. The default value is 200 seconds.

twaitsetup seconds: Configure the interval for waiting for the Setup message after the called party connects to a logical channel, ranging from 3 seconds to 30 seconds. The default value is 20 seconds.

Description

Use the voip h323-timer command to configure the timers in H323 protocol stack.

Use the undo voip h323-timer command to restore the default value.

Example

# Configure the socket timer to 28 seconds.

[VG-voice] voip h323-timer socket-create 28

1.1.115 vqa data-statistic

Syntax

vqa data-statistic { enable | disable }

View

Voice view

Parameter

enable: Enable statistics of voice data.

disable: Disable statistics of voice data.

Description

Use the vqa data-statistic command to enable/disable statistics of voice data.

By default, statistics of voice data is disabled.

In order to quickly locate trouble in VoIP call and perform debugging, use the vqa data-statistic enable command to enable statistics of voice data. Information to be performed statistics includes time for successfully searching the voice table, total number of received data packets, time for searching the table in fast and common modes and voice and fax of receive and transmit channels.

Statistics of voice data mainly serves debugging. Therefore, the user is recommended to disable this function when the service is normal so as to ensure higher performance of voice data processing.

Related commands: reset voice voip data-statistic, display voice voip data-statistic.

Example

# Enable the function of statistics for voice data.

[VG-voice] vqa data-statistic enable

1.1.116 vqa dsp-monitor

Syntax

vqa dsp-monitor buffer-time [ time ]

undo vqa dsp-monitor

View

Voice view

Parameter

buffer-time time: Time-length of the data to be buffered by DSP, in milliseconds. The value range is 180 to 480.

Description

Use the vqa dsp-monitor command to set a time-length for the DSP to buffer data.

Use the undo vqa dsp-monitor command to delete the monitoring time-length of the DSP to buffer data.

By default, data buffered by DSP is monitored for 270 milliseconds.

After you set a buffer-time, DSP automatically drops the expired voice data that it buffers to keep voice delay within an acceptable range.

You are recommended to set buffer-time to a value greater than 240 milliseconds. If this value is too small, bad voice quality will be delivered when large jitter occurs.

Example

# Set the DSP buffer-time to 270 milliseconds.

[VG-voice] vqa dsp-monitor buffer 270

1.1.117 vqa ip-precedence

Syntax

vqa ip-precedence tos-value

View

Voice view

Parameter

tos-value: The precedence level of voice signaling IP packets, which is an integer in the range of 0 to7 and defaults to 0. 0 represent the highest precedence level, and 7 represents the lowest precedence level.

Description

Use the vqa ip-precedence command to configure the precedence of all voice signaling IP packets (the ToS field of IP packets).

Use the undo ip-precedence command to restore the default value.

In practice, you should distinguish this command from the ip-precedence command in voice entity view. The latter is used only to configure the precedence of the entity-relevant voice or fax packets, while vqa ip-precedence in voice view is used to configure the precedence of all the voice signaling IP packets.

Related command: ip-precedence.

Example

# Set the precedence of all voice signaling IP packets to 5.

[VG-voice] vqa ip-precedence 5

1.1.118 vqa jitter-buffer

Syntax

vqa jitter-buffer depth

View

Voice view

Parameter

depth: Identify the Jitter Buffer depth in the range of 0 to 10. Setting Jitter Buffer depth to 0 indicates not to enable the Jitter Buffer function. The larger the value, the larger the depth, that is, the more complexly the Jitter Buffer processes the voice packets. By default, the Jitter Buffer depth is 3.

Description

Use the vqa jitter-buffer command to configure Jitter Buffer.

For networks of different performances, the depth for Jitter Buffer to provide the best performance is different. Therefore the Jitter Buffer depth must be selected based on a long-term observing and attempts, and in depth analysis of the actual network situations. So that the reasonable value fitting to the network requirements can be selected. If the depth value is too small, the Jitter Buffer function cannot provide the best loss compensation, disorder adjustment, jitter removing and re-discarding processing. If the depth value is too large, the delay will become influential to the receiving voice gateway.

Example

# Configure the depth of Jitter Buffer as 5.

[VG-voice] vqa jitter-buffer 5

1.1.119 vqa performance

Syntax

vqa performance { receive | send } { fast | normal }

View

Voice view

Parameter

fast: Enable fast receive and send process of voice data.

normal: Enable normal receive and send process of voice data, i.e., disable fast receive and send process.

Description

Use the vqa performance command to enable/disable fast receive and send process of voice data.

By default, the fast receive and send process of voice data is enabled..

Voice data forwarding is broken into two modes: normal and fast. Fast forwarding includes fast receive and fast send. The user can configure data performance switches as required.

Compared to normal receive process, fast receive process reduces the processes of memory application and data copy appropriately and improves receive speed of voice data. With the break mechanism, fast send process packs voice data and sends the packets to network layer according to the routing and link information to perform forwarding.

Example

# Disable fast receive and send process of voice data, i.e., enable normal receive and send process.

[VG-voice] vqa performance receive normal

[VG-voice] vqa performance send normal

Chapter 2 Fax Configuration Commands

2.1 Fax Configuration Commands

2.1.1 cngced-detection

Syntax

cngced-detection threshold times

undo cngced-detection

View

Voice subscriber line-view

Parameter

threshold: Detection threshold of CNG/CED signal when faxing in the range from 0 to 30 and the default value is 0.

times: Detection times threshold of CNG/CED signal when faxing in the range from 0 to 100 and the default value is 10.

Description

Use the cngced-detection command to set the threshold parameter for detecting CNG/CED signals.

Use the undo cngced-detection command to restore the default value.

Calling Tone (CNG) is generated when the fax at the calling end starts. Called Terminal Identification (CED) is generated when the fax at the called end starts.

A VG confirms the fax state by detecting the CNG/CED, however, as it is applied in various environments, sometimes it may fail to detect or have error detection. In these circumstances, you can use this command to set the detection parameter to adjust the sensation and reliability of device.

The greater the threshold and times values are, the more reliable the detection is; however, if the value is too big, some CNG/CED signals may be overlooked in detection. The times argument reflects the lower limit of the CNG/CED duration. For instance, the default value of times is 10, which means that a CNG/CED signal is regarded valid only if it lasts at least 300 ms. The minimum value of duration increases by 30 ms when the times value increases by 1.

& Note:

This function is applicable to VG 10-40 and VG 10-41 only.

Example

# Set the threshold to 5 and times to 20 in voice subscriber line interface for detecting CNG/CED signal.

[VG-voice-line0] cngced-detection 5 20

2.1.2 debugging voice fax

Syntax

debugging voice fax { all | api | channel channel-number | controller | error-all | ipp | t38 | cc }

View

Any view

Parameter

all: Enables all fax debugging of the Fax module.

api: Enables the API function debugging of the Fax module.

channel channel-number: Enables the debugging on a specified channel of the Fax module.

controller: Enables the controller debugging of the Fax module.

error-all: Enable the debugging of the Fax module on errors at all levels.

ipp: Enables the debugging on the information between the Fax and the IPP modules.

t38: Enables the T.38 information debugging of the Fax module.

cc: Enables the debugging for CC.

Description

Use the debugging voice fax command to enable debugging for fax.

The debugging voice fax channel channel-number command and the debugging voice fax all command are independent of each other. The debugging voice fax channel channel-number command enables debugging only for the specified channel, without giving the detailed debugging information. For the detailed debugging information, use the debugging voice fax combined with the specific keyword. The debugging voice fax all command enables debugging for all faxes except channels.

The undo debugging voice fax all command and the undo debugging voice fax channel command are not completely independent of each other. Performing the undo debugging voice fax all command can enable the undo debugging voice fax channel command at the same time. But the undo debugging voice fax channel command is only to disable the debugging for the channel.

Example

# Enable all the debugging of the Fax module.

[VG] debugging voice fax all

2.1.3 debugging voice vas fax

Syntax

debugging voice vas fax

View

Any view

Parameter

None

Description

Use the debugging voice vas fax command to enable the debugging on the fax data written and read between the VAS module and the voice card.

Example

# Enable the debugging on the fax data written and read between the VAS module and the voice card.

[VG] debugging voice vas fax

2.1.4 default entity fax

Syntax

default entity fax baudrate { 14400 | 2400 | 4800 | 9600 | disable | voice }

default entity fax ecm

default entity fax level level

default entity fax local-train threshold threshold

default entity fax nsf-on

default entity fax protocol { h323-t38 | pcm { g711alaw | g711ulaw } | t38 ]

default entity fax redundancy { hb-redundancy | lb-redundancy } number

default entity fax support-mode { rtp | vt }

default entity fax train-mode { local | ppp }

undo default entity fax { baudrate | ecm | level | local-train threshold | nsf-on | protocol | redundancy { hb-redundancy | lb-redundancy } | support-mode | train-mode }

View

Voice dial-program view

Parameter

The parameters of the default entity fax command include level, local-train, protocol, baudrate, nsf-on, support-mode, train-mode, and ecm. For the description of the parameters, see the appropriate fax baudrate, fax ecm, fax level, fax local-train threshold, fax nsf-on, fax protocol, fax redundancy, fax support-mode and fax train-mode commands.

Description

Use the default entity fax command to set the default fax parameter settings in the global scope.

Use the undo default entity fax command to restore the system default settings.

For the default setting of each parameter, see the default setting of the appropriate fax command.

You can configure the default setting of each fax parameter in the global scope by configuring the default entity fax command. In this case, all the existing voice entities that have not been configured with this function and all the new voice entities will inherit the configuration.

& Note:

When the negotiated voice compensation mode is G.711 and the fax rate fax baudrate is “disabled” (the fax forwarding capability is disabled), the command functions the same as the default entity fax protocol pcm command.

Example

# Set the default GW carrier level to -20 in the global scope.

[VG-voice-dial] default entity fax level -20

2.1.5 display voice fax statistics

Syntax

display voice fax statistics

View

Any view

Parameter

None

Description

Use the display voice fax statistics command to view the fax statistics information of the fax module.

& Note:

In the following two cases, the VG does not perform fax information statistics:

l When the VG is configured to work in the PCM passthrough mode

l When the negotiated voice compensation mode is G.711 and the fax rate fax baudrate is “disabled” (the fax forwarding capability is disabled), as in this mode this command functions the same as the default entity fax protocol pcm command

In addition, VG 20-16 and VG 20-32 do not support local fax information statistics.

Example

# Display the statistics of the Fax module.

[VG] display voice fax statistics

Statistics about fax Session:

{

Total : 0

FAX_PROTOCOL_T38 : 0

FAX_PROTOCOL_H323-T38 : 0

Success : 0

FAX_PROTOCOL_T38 : 0

FAX_PROTOCOL_H323-T38 : 0

Failure : 0

FAX_PROTOCOL_T38 : 0

FAX_PROTOCOL_H323-T38 : 0

Total fax time : 00:00:00

FAX_PROTOCOL_T38 : 00:00:00

FAX_PROTOCOL_H323-T38 : 00:00:00

Total processed pages : 0

FAX_PROTOCOL_T38 : 0

FAX_PROTOCOL_H323-T38 : 0

}

Statistics about using fax baudrate :

{

V27_2400 : 0

V27_4800 : 0

V29_7200 : 0

V29_9600 : 0

V17_7200 : 0

V17_9600 : 0

V17_12000: 0

V17_14400: 0

}

Statistics about using ECM or Non-ECM mode :

{

ECM : 0

Non-ECM: 0

}

Statistics about release reason :

{

WAIT_DP_BEG_DEMODULATE_TIMEOUT : 0

WAIT_DP_BEG_MODULATE_TIMEOUT : 0

WAIT_DP_END_DEMODULATE_TIMEOUT : 0

WAIT_DP_END_MODULATE_TIMEOUT : 0

WAIT_FRAMEACK_TIMEOUT : 0

WAIT_T30MSG_PSTN_TIMEOUT : 0

WAIT_T30MSG_IP_TIMEOUT : 0

SPOOL_TIME_OVER : 0

GET_INVALID_T30MESSAGE : 0

IPP_CALL_RELEASE : 0

NORMAL_RELEASE : 0

UNKNOWN_REASON : 0

}

Table 2-1 Description on the fields of the display voice fax command

Field	Description
Total	Total number of fax sessions
FAX_PROTOCOL_T38	Total number of fax sessions using T.38
FAX_PROTOCOL_H323-T38	Total number of fax sessions using H323-T38
Success	Total number of successful fax sessions
FAX_PROTOCOL_T38	Total number of successful fax sessions using T.38
FAX_PROTOCOL_H323-T38	Total number of successful fax sessions using H.323-T.38
Failure	Total number of failed fax sessions
FAX_PROTOCOL_T38	Total number of failed sessions using T.38 fax
FAX_PROTOCOL_H323-T38	Total number of failed fax sessions using H.323-T.38
Total Fax Time	Total time of fax sessions
FAX_PROTOCOL_T38	Total time of fax sessions using T.38
FAX_PROTOCOL_H323-T38	Total time of fax sessions using H.323-T.38
Processed Pages	Total number of processed fax pages
FAX_PROTOCOL_T38	Total number of fax pages processed using T.38
FAX_PROTOCOL_H323-T38	Total number of fax pages processed using H.323-T.38
V27_2400	Total number of fax sessions at 2400bps of V.27
V27_4800	Total number of fax sessions at 4800bps of V.27
V29_7200	Total number of fax sessions at 7200bps of V.29
V29_9600	Total number of fax sessions at 9600bps of V.29
V17_7200	Total number of fax sessions at 7200bps of V.17
V17_9600	Total number of fax sessions at 9600bps of V.17
V17_12000	Total number of fax sessions at 12000bps of V.17
V17_14400	Total number of fax sessions at 14400bps of V.17
ECM	Total number of fax sessions in ECM mode
Non-ECM	Total number of fax sessions in non-ECM mode
WAIT_DP_BEG_DEMODULATE_TIMEOUT	Times of fax release due to the expiration of the time waiting for the modem to start demodulation
WAIT_DP_BEG_MODULATE_TIMEOUT	Times of fax release due to the expiration of the time waiting for the modem to start modulation
WAIT_DP_END_DEMODULATE_TIMEOUT	Times of fax release due to the expiration of the time waiting for the modem to stop demodulation
WAIT_DP_END_MODULATE_TIMEOUT	Times of fax release due to the expiration of the time waiting for the modem to stop modulation
WAIT_FRAMEACK_TIMEOUT	Times of fax release due to the expiration of the time waiting for V.21 data transmission
WAIT_T30MSG_PSTN_TIMEOUT	Times of fax release due to the expiration of the time waiting for the T.30 signal from the fax machine
WAIT_T30MSG_IP_TIMEOUT	Times of fax release due to the expiration of the time waiting for the remote T.30 signal
SPOOL_TIME_OVER	Times of fax release due to the expiration of the fax spoofing mechanism
GET_INVALID_T30MESSAGE	Times of fax release due to the receipt of invalid T.30 messages
IPP_CALL_RELEASE	Times of fax release due to session termination
NORMAL_RELEASE	Times of normal fax release
UNKNOWN_REASON	Times of fax release due to unknown reasons

2.1.6 fax baudrate

Syntax

fax baudrate { 14400 | 2400 | 4800 | 9600 | disable | voice }

undo fax baudrate

View

Voice entity view

Parameter

14400: First negotiates according to the V.17 fax protocol, with the allowed maximum fax speed being 14400 bps.

2400: Preferred fax speed is 2400 bps.

4800: First negotiates according to the V.27 fax protocol, with the allowed maximum fax speed being 4800 bps.

9600: First negotiates according to the V.29 fax protocol, with the allowed maximum fax speed being 9600 bps.

disable: Disables the fax forwarding function.

voice: First determines the allowed maximum fax speed depending on the adopted voice codec protocol.

Description

Use the fax baudrate command to configure the maximum fax speed that the GW allows.

Use the undo fax baudrate command to restore the default maximum fax speed.

By default, fax speed is determined in the voice mode.

The command applies to IP Fax.

If the rate is set to be some other value other than “disable” and “voice”, rate negotiation will be performed first according to the fax protocol specifying the rate. The rate configured here is the allowed fastest rate rather than the one actually used.

If the rate is set to voice, the allowed maximum fax rate will be determined depending on the actually adopted voice codec protocol. Due to the network bandwidth limitation, like voice encoding/decoding, the bandwidth also needs to be considered in the fax rate configuration. To ensure normal fax operation, it is recommended that you adopt the following configuration methods:

l If G.711 applies, the maximum fax rate and fax protocol will be 14400 bps and V.17;

l If G.723.1 Annex A applies, the maximum fax rate and fax protocol will be 4800 bps and V.27;

l If G.729 applies, the maximum fax rate and fax protocol will be 9600 bps and V.29.

If “disable” is selected, the fax function will be disabled.

& Note:

Example

# Apply V.29 on the VG for fax rate negotiation.

[VG-voice-dial-entity4] fax baudrate 9600

2.1.7 fax ecm

Syntax

fax ecm

undo fax ecm

View

Voice entity view

Parameter

None

Description

Use the fax ecm command to apply the ECM mode on the VG by force.

Use the undo fax ecm command to disable the ECM mode on the VG.

By default, ECM mode is disabled on the VG.

This command is used for performing the forced limitation at the VG. If the fax machines at both ends support the ECM mode but the non-ECM mode has been configured at the VG, then the non-ECM mode is selected. If one or two of the fax machines at both ends do not support the ECM mode, the non-ECM mode is selected. Only when the fax terminals at both ends support the ECM mode and the VG adopts the ECM mode will the ECM mode be adopted.

Example

# Apply the ECM mode on the VG by force.

[VG-voice-dial-entity4] fax ecm

2.1.8 fax level

Syntax

fax level level

undo fax level

View

Voice entity view

Parameter

level: GW carrier level, namely the attenuation value in dBm of the sending level. The value range is -60 to -3 and the default value is to -15. The bigger the level value is, the stronger the energy is; the smaller the level value is, the greater the attenuation is.

Description

Use the fax level command to configure a GW carrier level.

Use the undo fax level command to restore the default GW carrier level.

This command applies to IP Fax.

Normally, the default GW carrier level is acceptable. In the event that the system fails in setting up a session when all the other configurations are correct to try to adjust the GW carrier level.

Example

# Set the GW carrier level to -20.

[VG-voice-dial-entity4] fax level -20

2.1.9 fax local-train threshold

Syntax

fax local-train threshold threshold

undo fax local-train threshold

View

Voice entity view

Parameter

threshold: Local training threshold percentage in the range of 0 to 100. By default, it is set to 10.

Description

Use the fax local-train threshold command to configure the local fax training threshold percentage.

Use the undo fax local-train threshold command to restore the default local fax training threshold percentage.

This command applies to IP Fax.

When two fax machines make rate training, the sender first sends the TCF data in the form of “0” to the receiver for 1.5 (±10%) seconds, and the receiver determines whether the current rate is acceptable based on the received TCF data.

In the event that the training mode is set to local training, this command can be used to configure the local training threshold. If errors occur in the data transmission, the received TCF data will contain “1”(s). If the percentage of the number of these “1”s to all the received TCF data is less than the configured “threshold”, the current speed training is successful. Otherwise, the current training fails.

Example

# Set the local fax training threshold percentage to 20.

[VG-voice-dial-entity4] fax local-train threshold 20

2.1.10 fax nsf-on

Syntax

fax nsf-on

undo fax nsf-on

View

Voice entity view

Parameter

None

Description

Use the fax nsf-on command to enable the non-standard mode for the transmission of fax facilities signals.

Use the undo fax nsf-on command to restore the default setting of fax facilities signal transmission.

By default, the standard mode is used for fax facilities signal transmission.

This command only applies to IP Fax.

In some occasions, when transmitting encrypted faxes for example, the Non-Standard Facilities (NSF) mode is rather important for the fax communications. It is necessary to configure the fax nsf-on command, so that the fax terminals at both sides can start transmission with exchanging NSF messages and then complete the subsequent fax negotiation and communicate with each other. NSF messages are standard T.30 messages carrying private properties.

Example

# Enable the NSF mode for fax facilities signal transmission.

[VG-voice-dial-entity4] fax nsf-on

2.1.11 fax protocol

Syntax

fax protocol { h323-t38 | pcm { g711alaw | g711ulaw } | t38 }

undo fax protocol

View

Voice entity view

Parameter

h323-t38: Standard H323 T.38 fax protocol. This parameter is used for interoperation with the Cisco T.38 fax protocol

pcm: VG fax passthrough protocol, using the PCM mode.

g711alaw: Adopts G.711 A-law voice codec mode for fax passthrough transmission.

g711ulaw: Adopts G.711 µ-law voice codec mode for fax passthrough transmission.

t38: Adopts T.38 fax protocol, which enables fast fax connection establishment.

Description

Use the fax protocol command to configure the protocol for the interoperation with other devices.

Use the undo fax protocol command to restore the default fax protocol.

This command only applies to IP Fax.

The VGs support the standard packet format defined by ITU-T T.38.

The VGs support two fax starting modes: H.323 negotiation mode and VG negotiation mode, corresponding to fax protocols H.323-T.38 and T.38 respectively.

The VGs support fax passthrough transmission through the following two types of configurations:

l The PCM mode is enabled as the fax protocol.

l The negotiated voice codec mode is G.711, and the fax baudrate is set to “disable” (fax forwarding is disabled), and the VAD is disabled to avoid fax failure. This mode is applicable to passthrough transmission with other devices.

Example

# Set the number of transmitted redundant high-speed packets in the T.38 protocol to 2.

[VG-voice-dial-entity4] fax protocol t38 hs-redundancy 2

2.1.12 fax redundancy

Syntax

fax redundancy { hb-redundancy | lb-redundancy } number

undo fax redundancy { hb-redundancy | lb-redundancy }

View

Voice entity view

Parameter

hb-redundancy number: Number of high baudrate redundant packets. It ranges from 0 to 2 and the default value is 0.

lb-redundancy number: Number of low baudrate redundant packets. It ranges from 0 to 5 and the default value is 0.

Description

Use the fax redundancy command to set the number of high baudrate and low baudrate redundant packets.

Use the undo fax redundancy command to restore the default number of redundant packets.

This command only applies to fax protocol in the configuration of H323-t38 or t38.

Example

# Set the number of high baudrate redundant packets to 2.

[VG-voice-dial-entity88]fax redundancy hb-redundancy 2

2.1.13 fax support-mode

Syntax

fax support-mode { rtp | vt }

undo fax support-mode

View

Voice entity view

Parameter

rtp: Uses the rtp mode.

vt: Uses the vt mode (for interoperation with a VocalTec GW).

Description

Use the fax support-mode command to configure the fax transmission mode.

Use the undo fax support-mode command to restore the default fax transmission mode.

By default, the rtp mode is adopted.

This command only applies to IP Fax.

In normal cases, the rtp mode is adopted, but the vt mode should be adopted when the VG is communicating with a VocalTec GW.

Example

# Adopt the vt mode as the fax transmission mode.

[VG-voice-dial-entity4] fax support-mode vt

2.1.14 fax train-mode

Syntax

fax train-mode { local | ppp }

undo fax train-mode

View

Voice entity view

Parameter

local: Adopts the local training mode.

ppp: Adopts the peer-to-peer training mode.

Description

Use the fax train-mode command to configure a training mode for the VG.

Use the undo fax train-mode command to restore the default training mode on the VG.

By default, PPP mode is adopted.

This command applies to IP Fax.

The rate training between the fax machines that has the participation of the GWs at two ends is called local training. In this mode, the training is first implemented between the fax machine and the attached GW at each end. Then, the receiving GW sends the training result of the receiver to the GW of the sender, and the sending GW will determine the ultimate packet transmission rate by comparing the training results of both ends.

In the peer-to-peer training mode, however, the GWs do not participate in the rate training between the fax machines at both ends. In this mode, the rate training is conducted between the two fax terminals and is in a pass-through mode from the perspective of the GWs.

Example

# Apply the local training mode on the VG.

[VG-voice-dial-entity4] fax train-mode local

2.1.15 reset voice fax

Syntax

reset voice fax

View

System view

Parameter

None

Description

Use the reset voice fax command to clear the fax statistics information.

Example

# Clear the fax statistics information.

[VG] reset voice fax

Chapter 3 E1 Voice Configuration Commands

3.1 E1 Voice Configuration Commands

3.1.1 ani

Syntax

ani

undo ani

View

R2 CAS view

Parameter

None

Description

Use the ani command to enable the terminating point to send the calling party information (service category and calling number) to the originating point during call connecting process.

By default, the terminating point does not send the calling party information to the originating point.

Configure the local end with this command to support the automatic number identification.

Related command: cas and ani-offset.

Example

# Configure the local exchange to require the opposite exchange to send the calling number during the connecting process.

[VG-cas0:0] ani

3.1.2 ani-offset

Syntax

ani-offset number

undo ani-offset

View

R2 CAS view

Parameter

number: The quantity of the collected numbers, with a value ranging from 1 to 10. By default, the number is 1.

Description

Use the ani-offset command to configure the number of digits of called numbers to be collected before requesting the calling party information.

Use the undo ani-offset command to restore the default value.

This command is used for setting the number of the digits to be collected before requesting the calling number or calling identifier. When the quantity of the collected numbers is less than this value, the system will wait to the next number till timeout, and in the waiting it will not request the calling number information from the remote end. When the quantity of the collected numbers is equal to or exceeds this value, it is able to request the calling number or calling identifier from the remote end.

By default, the ani-offset command takes effect after the execution of the ani command.

Related command: cas, timeouts, reverse, and renew.

Example

# Set to start requesting the calling number or calling identifier after receiving the 3-digit called numbers.

[VG-cas0:0] ani-offset 3

3.1.3 answer

Syntax

answer { enable | disable }

View

R2 CAS view

Parameter

enable: Enable the terminating point to send answer signals so that both parties can start conversation only after the answer signal is received.

disable: Disable the terminating point to send answer signals. In this case, the terminating point is not required to send an answer signal; the originating point reports an answer message automatically to the upper layer after the timer expires and both parties begin conversation.

Description

Use the answer command to configure the terminating point to send answer signals.

By default, the originating point is enabled to send answer signals.

In some countries, answer signals are not sent in the R2 line signaling coding scheme. In this case, you can use the answer command to adapt this signaling coding scheme. If the originating point does not require the terminating point to answer signals, the terminating point confirms call connection after the specified time.

Related commands, refer to re-answer and timer dl re-answer.

Example

# Disable the terminating point to send answer signals.

[VG-cas0:0] answer disable

3.1.4 callmode

Syntax

callmode { segment | terminal }

undo callmode

View

R2 CAS view

Parameter

segment: Indicates the segment to segment mode. In this mode, one party answers directly the kb signal after receiving it and then initiates a call to the IP side.

terminal: Indicates terminal to terminal mode. In this mode, one party calls IP side after receiving kd signal. Only after acquiring the called status, it sends kb signal to the central office.

Description

Use the callmode command to configure the connection mode of the call using R2 signaling.

Use the undo callmode command to restore the default value of the connection mode.

By default, the terminal mode is applied.

Example

# Configure the segment command for the call connection using R2 signaling.

[VG-cas0:0] callmode segment

3.1.5 cas

Syntax

cas ts-set-number

View

CE1/PRI interface view

Parameter

ts-set-number: The serial number of the predefined Timeslot set, ranging from 0 to 30.

Description

Use the cas command to enter the R2 CAS and digital E&M signaling view.

The command cas is used to enter the customized R2 CAS view. Under this mode various parameters of the R2 signaling on the E1 interface can be configured as required. To validate the customized R2 parameters, it is required to keep the parameter of ts-set-number in the command cas consistent with the parameter set-number in the command timeslots-set.

Related command: timeslots-set, ani-offset, effect-time, reverse, select-mode, timer, trunk-direction, and renew.

Example

# Enter the R2 CAS view of No.5 Timeslot set.

[VG-E1-0] timeslot-set 5 timeslot-list 1-31 signal r2

[VG-E1-0] cas 5

3.1.6 clear-forward-ack

Syntax

clear-forward-ack { enable | disable }

View

R2 CAS view

Parameter

enable: Enables the terminating point to acknowledge the clear-forward signal by sending a clear-back signal.

disable: Disables the terminating point to acknowledge the clear-forward signal by sending a clear-back signal.

Description

Use the clear-forward-ack command to enable or disable the terminating point to respond by sending a clear-back signal when the originating point (calling party) disconnects a call.

By default, the terminating point does not send a clear-back signal to acknowledge the clear-forward signal.

In some countries, if the terminating point controls the relay circuit reset in the R2 signaling exchange process, when the calling party disconnects a call and the originating point sends a clear-forward signal to the terminating point, the terminating point sends a clear-back signal as an acknowledgement, and then sends a release guard signal to indicate that the line of the terminating point is thoroughly released.

Related command: mode china default-standard.

Example

# Enable the terminating point to send a clear-back signal to acknowledge the clear-forward signal.

[VG-cas0:0] clear-forward-ack enable

3.1.7 clock

Syntax

clock { master | slave }

View

CE1/PRI interface view

Parameter

master: Master clock is used.

slave: Slave clock is used.

Description

Use the clock command to set the clock mode on the CE1/PRI interface.

Use the undo clock command to restore the default clock mode.

By default, the slave mode is used on the interface.

When the CE1/PRI interface is used as the DCE device, use the master mod. If the interface is used as the DTE device, choose the slave mode.

Example

# Configure to choose the master mode on CE1/PRI interface.

[VG-E1-0] clock master

3.1.8 code

Syntax

code { ami | hdb3 }

View

CE1/PRI interface view

Parameter

ami: Indicates the Alternate Mark Inversion (AMI) coding format.

hdb3: Indicates the High Density Bipolar 3 (hdb3) coding format. It is only effective for the CE1/PRI interface.

Description

Use the code command to configure the coding format on CE1/PRI interface.

Use the undo code command to restore the default format.

By default, hdb3 is used on CE1/PRI interface.

Note that the coding format must be the same as that on the peer device,

Example

# Set the AMI coding format on the interface E1 0.

[VG-E1-0] code ami

3.1.9 controller

Syntax

controller e1 e1-number

View

System view

Parameter

e1-numbe: CE1/PRI interface number.

Description

Use the controller e1 command to enter the CE1/PRI interface view.

Example

# Enter the E1 0 interface view.

[VG] controller e1 0

[VG-E1-0]

3.1.10 debugging voice r2

Syntax

debugging voice r2 { all | ccb controller e1-number timeslots | dl | dtmf | error | mfc | msg | rcv | warning }

View

Any view

Parameter

all: Enables all the debugging in R2 software module.

ccb controller: Enables all the CCB debugging in R2 signaling.

dl: Enables the line signaling debugging in R2 signaling.

dtmf: Enables the Dual-Tone Multifrequency (DTMF) debugging in R2 signaling.

error: Enables the error debugging in R2 software module.

msg: Enables the message interface debugging in R2 signaling.

mfc: Enables the interregister signaling debugging in R2 signaling.

rcv: Enables the RCV software module debugging in R2 software module.

warning: Enables warning debugging in R2 software module.

e1-number: E1 port number.

timeslots: Time slot number, ranging from 1 to 31.

Description

Use the debugging voice r2 command to enable the corresponding debugging in R2 signaling module.

Use the debugging voice r2 ccb command to view the information of the corresponding control block by specifying the E1 port number and time slot number.

Example

# Enable all the line signaling debugging in R2 signaling on E1 voice port.

[VG] debugging voice r2 dl

3.1.11 debugging voice rcv r2

Syntax

debugging voice rcv r2

View

Any view

Parameter

r2: Enable the debugging between the RCV module and the R2 module of bottom layer.

Description

Use the debugging voice rcv r2 command to enable the debugging between the RCV module and the R2 module of bottom layer.

Example

None

3.1.12 debugging voice vpp r2

Syntax

debugging voice vpp r2

View

Any view

Parameter

r2: Enable the debugging between the VPP module and the R2 module of bottom layer.

Description

Use the debugging voice vpp r2 command to enable the debugging between the VPP module and the R2 module of bottom layer.

Example

None

3.1.13 default

Syntax

default

View

R2 CAS view

Parameter

None

Description

Use the default command to restore the default values of all the R2 configurations.

The following table gives the configuration items affected by this command.

Table 3-1 Configuration items affected by the default command

Configuration item	Default
The originating exchange sends the seizure acknowledgement signal.	ENABLE
The originating exchange sends the answer signal.	ENABLE
The originating exchange answers at the time of clear-back.	DISABLE
The originating exchange answers with the clear-back signal at the time of clear-forward.	DISABLE
Answer signal timeout interval	60 seconds
Clear-forward signal timeout interval	10 seconds
Seizure signal timeout interval	1 second
Re-answer timeout interval	1 second
Delay before sending the release-guard signal upon a timeout	10 second
Seizure acknowledgement signal timeout interval	40 seconds
The ABCD bit pattern that represents an idle receive line	1001
The ABCD bit pattern that represents an idle transmit line	1001
The ABCD bit pattern that represents a seized receive line	0001
The ABCD bit pattern that represents a seized transmit line	0001
The ABCD bit pattern that represents the seizure acknowledged state of the transmit line.	1101
The ABCD bit pattern that represents the seizure acknowledged state of the receive line.	1101
The ABCD bit pattern that represents the answered state of the receive line.	0101
The ABCD bit pattern that represents the answered state of the transmit line.	0101
The ABCD bit pattern that represents the clear-forward state of the receive line.	1001
The ABCD bit pattern that represents the clear-forward state of the transmit line.	1001
The ABCD bit pattern that represents the clear-back state of the receive line.	1101
The ABCD bit pattern that represents the clear-back state of the transmit line.	1101
The ABCD bit pattern that represents the release-guard state of the receive line.	1001
The ABCD bit pattern that represents the release-guard state of the transmit line.	1001
The ABCD bit pattern that represents a blocked receive line.	1101
The ABCD bit pattern that represents a blocked transmit line.	1101
Country mode	ITU-T default

Example

# Restore the default of all the configurations on CAS 0:5.

[VG-E1-0] timeslot-set 5 timeslot-list 1-31 signal r2

[VG-cas0:5] default

3.1.14 delay

Syntax

delay { call-interval | hold | rising | send-dtmf | dtmf | dtmf-interval | wink-rising | wink-hold | send-wink } milliseconds

undo delay { call-interval | hold | rising | send-dtmf | dtmf | dtmf-interval | wink-rising | wink-hold | send-wink }

View

Digital E&M voice subscriber-line view

Parameter

call-interval milliseconds: The call interval, set on the digital E&M subscriber-line, is in the range of 200 to 2000 ms. By default, the value is 200ms.

hold milliseconds: The maximum duration the caller will wait for the callee to hang up so as to send the DTMF number in digital E&M subscriber-line delay-start mode. It ranges from 100 to 5000ms, and the default value is 400ms.

rising milliseconds: The maximum duration the caller will wait from the sending of off-hook signal to the callee's status detected in digital E&M subscriber-line delay-start mode. It ranges from 20 to 2000ms, and the default value is 300ms.

send-dtmf milliseconds: The delay before the caller sends DTMF number in digital E&M subscriber-line immediate-start mode. It ranges from 50 to 5000ms, and is defaulted to 300ms.

dtmf milliseconds: The lasting duration of the DTMF signals set on the digital E&M subscriber-line in the range of 50 to 500ms. By default, the value is 120ms.

dtmf-interval milliseconds: The interval between two DTMF signals set on the digital E&M subscriber-line in the range of 50 to 500ms. By default, the value is 120ms.

wink-rising milliseconds: The maximum duration the caller will wait for the wink signals after sending the seizure signals in digital E&M subscriber-line wink-start mode, set on the called end. It ranges from 100 to 5000ms, and is defaulted to 2000ms.

wink-hold milliseconds: The maximum lasting duration of the wink signals are received by the caller in digital E&M subscriber-line wink-start mode. It ranges from 100 to 3000ms, and is defaulted to 500ms.

send-wink milliseconds: Specify how long the called party will delay sending wink signal after receiving the seizure signal on digital E&M interface. It ranges from 100 to 5000ms and defaults to 200ms.

Description

Use the delay command to configure the related time parameters at the digital E&M subscriber-line (E1 controller).

Use the undo delay command to restore these parameters to the default value.

On the digital E&M subscriber-line, the waiting duration of originating the next call set by the delay call-interval milliseconds command is applicable to the immediate, wink and delay start.

When the digital E&M subscriber-line uses the delay start mode, the calling party will use the delay hold milliseconds command to set the longest time waiting for the delay signals .

When the digital E&M subscriber-line uses the delay start mode, use the delay rising milliseconds command to set the time waiting for the delay signals of called party after the calling party sends out the off-hook signals, and then detect the device state of the called party.

When the digital E&M subscriber-line uses the immediate start mode, use the delay send-dtmf milliseconds command to set the delay time before the calling party sends the called number.

The delay dtmf milliseconds command is used to set the duration that the DTMF signals are being sent, and the delay dtmf-interval milliseconds command is used to set the interval for sending the DTMF signals.

When the digital E&M subscriber-line uses the wink start mode, the delay wink-rising milliseconds command is used to set the longest time it takes for the calling party to wait for the wink signals after it sends the seizure signals.

When the digital E&M subscriber-line uses the wink start mode, the delay wink-hold milliseconds command is used to set the longest duration that the wink signals sent by the called party.

When the digital E&M subscriber-line uses the wink start mode, the delay send-wink milliseconds command is used to set the longest time that the called party delays before sending the wink signals.

Related command: timeslots-set and timer.

Example

# Set the longest delay-waiting duration to 3000 milliseconds.

[VG-voice-line1:3] timing delay hold 3000

3.1.15 dialtone-generate

Syntax

dialtone-generate

undo dialtone-generate

View

Digital E&M voice subscriber-line view

Parameter

None

Description

Use the dialtone-generate command to configure the VG as the terminating point to send the dial tone to the originating point in the preparation state of receiving numbers, prompting the calling party to dial the number.

Use the undo dialtone-generate command to configure the VG as the terminating point not to send the dial tone to the originating point in the preparation state of receiving numbers.

By default, the VG does not send the dial tone to the originating point.

Example

# Configure the digital E&M voice subscriber-line as the terminating point not to send the dial tone to the originating point.

[VG-voice-line1:3] undo dialtone-generate

3.1.16 display voice r2 call-statistics

Syntax

display voice r2 call-statistics

View

Any view

Parameter

None

Description

Use the display voice r2 call-statistics command to view the R2 call statistics.

Related command: reset voice r2.

Example

# Display the information of R2 signaling call statistics.

[VG] display voice r2 call-statistics

[ E1-Group(0:0) Call Statistics ]

+-------------------------------------------------+

[Call sumcount] -> 4

[call success] -> 4

[Call failure] -> 0

[Call-in count] -> 1

[Call-in success] -> 1

[Call-in failure] -> 0

[Call-in answer] -> 0

[Call-in nullnum] -> 0

[Call-in process] -> 0

[Call-out count] -> 3

[Call-out success] -> 3

[Call-out failure] -> 0

[Call-out answer] -> 3

[Call-out busy] -> 0

[Call-out nullnum] -> 0

[Call-out congestion] -> 0

[Call-out process] -> 0

Table 3-2 Description on the fields of the display voice r2 call-statistics command

Item	Description
Call sumcount	Total number of calls: total number of calls in E1 time slot group (sum of incoming and outgoing calls)
call success	Number of successful calls: total number of successful signaling connection, that is, R2 signaling connection is successfully completed, and the opposite exchange is available for another call connection.
Call failure	Number of failed calls: total number of failed signaling exchange during call connecting process, such as peer subscriber line busy, null called number, and line failure, etc.)
Call-in count	Total number of incoming calls
Call-in success	Total number of successful incoming calls
Call-in failure	Total number of failed incoming calls
Call-in answer	Number of sent answers: number of answer signals that the local exchange sends to the originating point when the call is connected successfully and the called party picks up the phone.
Call-in nullnum	Routing failure times: number of failed incoming calls due to no corresponding route available for the called number to create connection.
Call-in process	Call-in connection times
Call-out count	Total number of outgoing calls
Call-out success	Total number of successful outgoing calls
Call-out failure	Total number of failed outgoing calls
Call-out answer	Number of received answers: number of answer signals received from the terminating point when the call is connected successfully.
Call-out busy	Subscriber line busy times: number of subscriber line busy signals received from the terminating point during call connecting process.
Call-out nullnum	Count of null calling numbers: number of null number signals received from the terminating point during call connecting process.
Call-out congestion	Number of received congestion: number of congestion signals received from the terminating point during call connecting process.
Call-out process	Call-out connection times

3.1.17 display voice rcv statistic r2

Syntax

display voice rcv statistic r2

View

Any view

Parameter

None

Description

Use the display voice rcv statistic r2 command to view the information of call statistics related to the R2 signaling in the RCV module.

This command is used to display the interactive messages between the RCV module and the R2 signaling module, including the number of sending the messages of the connection request to acknowledge success and fail, the number of sending the messages of activation acknowledgement of success and failure, the number of sending the messages of on-hook and off-hook, the number of receiving the messages of connection request, the number of receiving activation messages, and the number of receiving such messages as release, ringing, and unknown.

Example

# Display the information of R2 signaling call statistics in RCV module .

[VG] display voice rcv statistic r2

Statistic between RCV and R2 :

{

Send_R2_ConnectReqAck_SUCCESS : 0

Send_R2_ConnectReqAck_FAIL : 0

Send_R2_ActiveAck_SUCCESS : 0

Send_R2_ActiveAck_FAIL : 0

Send_R2_Onhook : 0

Send_R2_Offhook : 0

Send_R2_IPAlerting : 0

Recv_R2_ConnectReq : 0

Recv_R2_Active_TD_IN : 0

Recv_R2_Active_TD_OUT : 0

Recv_R2_Active_ELSE : 0

Recv_R2_Release : 0

Recv_R2_Alert_AP_ALERTING : 0

Recv_R2_Alert_ELSE : 0

Recv_R2_Unknow : 0

}

Table 3-3 Description on the fields of the display voice rcv statistic r2 command

Item	Description
Send_R2_ConnectReqAck_SUCCESS	Number of sending connection request success-acknowledge message to R2 module
Send_R2_ConnectReqAck_FAIL	Number of sending connection request failure-acknowledge message to R2 module
Send_R2_ActiveAck_SUCCESS	Number of sending activation success-acknowledge message to R2 module
Send_R2_ActiveAck_FAIL	Number of sending activation failure-acknowledge message to R2 module
Send_R2_Onhook	Number of sending onhook message to R2 module
Send_R2_Offhook	Number of sending offhook message to R2 module
Send_R2_IPAlerting	Number of sending IP side alerting message to R2 module
Recv_R2_ConnectReq	Number of receiving R2 connection request message
Recv_R2_Active_TD_IN	Number of receiving R2 called end hooking off message
Recv_R2_Active_TD_OUT	Number of receiving R2 caller end hooking off message
Recv_R2_Active_ELSE	Number of receiving R2 other hooking off message
Recv_R2_Release	Number of receiving R2 release request message
Recv_R2_Alert_AP_ALERTING	Number of receiving R2 alerting message
Recv_R2_Alert_ELSE	Number of receiving R2 other alerting message
Recv_R2_Unknow	Number of receiving R2 unknown message

3.1.18 display voice subscriber-line

Syntax

display voice subscriber-line e1-number: { ts-set-number | 15 }

View

Any view

Parameter

e1-number: Indicate the number of subscriber line generated in creating the Timeslot set or the ISDN PRI set.

ts-set-number: Indicate the number of the Timeslot set created successfully.

Description

Use the display voice subscriber-line command to view the subscriber line configuration.

The command display voice subscriber-line e1-number:ts-set-number is mainly used to display the Timeslot set corresponding to the E1 subscriber line, whether to adopt the private-line auto-ring connection and connection number, the subscriber line description, whether to start the echo cancellation function and echo cancellation sampling time length, and whether to start the comfort noise function.

The command display voice subscriber-line e1-number:15 is mainly used to display the configuration information of the subscriber line corresponding to the ISDN PRI set, such as whether to adopt the private-line auto-ring connection, subscriber line description, whether to start the echo cancellation function and echo cancellation sampling time length, input gain and output attenuation, whether to adopt the nonlinear processing in echo cancellation, the time of waiting for the initial number and the dial-up time interval between numbers, etc.

Example

# Display the configuration of voice subscriber-line.

[VG] display voice subscriber-line 0:0

The voice line was ds0

this subscriber line was not set connection

The subscriber line's descrition:

echo cancellation enable

echo cancellation coverage 16

comfort noise enable

PCM companding type :A-law

[VG] display voice subscriber-line 1:15

The voice line was pri

this subscriber line was not set connection

The subscriber line's descrition:

echo cancellation enable

echo cancellation coverage 16

music threshold is -38

receive gain 0

transmit gain 0

first-dial timer 10

dial-interval Timer 10

PCM companding type :A-law

Table 3-4 Description on the fields of the display voice subscriber-line command

Item	Description
The voice line was	Signaling type of voice subscriber line
This subscriber line	Connection type of voice subscriber line
The subscriber line's description	Description of the voice subscriber line
echo cancellation	Configuration of echo cancellation on voice subscriber line
echo cancellation coverage	Coverage of echo cancellation on voice subscriber line
comfort noise	Configuration of comfort noise on voice subscriber line
PCM companding type	PCM companding type on voice subscriber line
music threshold	Music threshold on voice subscriber line
receive gain	Receive gain on voice subscriber line
transmit gain	Transmit gain on voice subscriber line
first-dial timer	Timeout of the local first calling on voice subscriber line
dial-interval Timer	Timeout of the local dial interval on voice subscriber line

3.1.19 display voice voip

Syntax

display voice voip { down-queue e1v1-no | phy-statistic e1vi-bno | up-queue e1vi-no | data-statistic { brief | channel | verbose } }

View

Any view

Parameter

e1vi-bno; Indicates the number of the E1V1 interface.

brief: Displays the brief statistics information of voice packets.

channel: Displays the statistics information of voice packets of a specified channel.

verbose: Displays the detailed information.

Description

Use the display voice voip downqueue e1vi-no command to display the contents of the down interrupt queue between the E1V1 interface and VG mainboard.

Use the display voice voip up-queue e1vi-no command to display the contents of the up interrupt queue between the E1V1 interface and VG mainboard.

Use the display voice voip phy-statistic e1vi-no command to display the statistics information of physical layer.

Use the display voice voip data-statistic command to display the statistics information of voice packets. Before this, use the vqa data-statistic enable command to enable the data statistics related to voice quality.

Example

# Display the contents of the down interrupt queue between the E1V1 interface and VG mainboard.

[VG] display voice voip down-queue 0

V = 0,I = 0,P = 0,C = 0,E = E1VI_NULL_EVENT, B = 0

V = 0,I = 1,P = 0,C = 0,E = E1VI_NULL_EVENT, B = 0

V = 0,I = 2,P = 0,C = 0,E = E1VI_NULL_EVENT, B = 0

……

V = 0,I = 255,P = 0,C = 0,E = E1VI_NULL_EVENT, B = 0

E1VI board 0 down interrupt queue is empty :

Table 3-5 Description on the fields of the display voice voip command

Item	Description
V	Value flag of interrupt
I	Sequence number of interrupt down-queue
P	Port number of E1VI board
C	Channel number of E1VI board
E	Type of event
B	Flag of queue ending
E1VI board 0 down interrupt queue is empty	Interrupt down-queue of E1VI interface is empty

3.1.20 dl-bits

Syntax

dl-bits { answer | blocking | clear-back | clear-forward | idle | seizure | seizure-ack | release-guard } rx-bits ABCD tx-bits ABCD

undo dl-bits { answer | blocking | clear-back | clear-forward | idle | seizure | seizure-ack | release-guard }

View

R2 CAS view

Parameter

answer: Answer signal of R2 line signaling.

blocking: Blocking signal of R2 line signaling.

clear-back: Clear-back signal of R2 line signaling.

clear-forward: Clear-forward signal of R2 line signaling.

idle: Idle signal of R2 line signaling.

seizure: Seizure signal of R2 line signaling.

seizure-ack: Seizure acknowledgement signal of R2 line signaling.

release-guard: Release guard signal of R2 line signaling.

rx-bits ABCD: Value of signal bits of receiving R2 line signaling, ranging from 0000 to 1111.

tx-bits ABCD: Value of signal bits of transmitting R2 line signaling, ranging from 0000 to 1111.

Table 3-6 Default value of the signals of R2 line signaling

Signal	rx-bits ABCDDefault value	tx-bits ABCDDefault value
Answer	0101	0101
Blocking	1101	1101
Clear-back	1101	1101
Clear-forward	1001	1001
Idle	1001	1001
Seize	0001	0001
Seizure-ack	1101	1101
Release-guard	1001	1001

Description

Use the dl-bits command to configure the bit value of all the signals of R2 line signaling.

Use the undo dl-bits command to restore the default value.

You can use the dl-bits command to configure the ABCD bits value of R2 signaling for different coding scheme in different countries.

Related commands: seizure-ack { enable/disable } and answer { enable/disable }.

Caution:

The line needs to be reset after the change of line signaling value (use the shutdown command to shut down the E1 interface, and then use the undo shutdown command to enable the E1 interface, or reconnect the line after the disconnection). Because the change of line signaling may affect the line state, do not configure the signaling during the conversation (using the R2 line).

Example

# Configure ABCD bits value of the idle signal of receiving R2 signaling to 1101, and that of the idle signal of transmitting R2 signaling to 1011.

[VG-cas0:0] dl-bits idle receive 1101

[VG-cas0:0] dl-bits idle transmit 1011

3.1.21 dtmf

Syntax

dtmf { enable | disable }

View

R2 CAS view

Parameter

enable: Enables R2 signaling to be received and sent in DTMF mode.

disable: Disables R2 signaling to be received and sent in DTMF mode.

Description

Use the dtmf command to configure the mode for sending and receiving R2 signaling.

By default, MFC mode, not DTMF mode, is used to collect call number information.

The dtmf command is used to configure whether to use MFC or DTMF in sending and receiving R2 signaling. DTMF mode is used when you configure dtmf enable. MFC mode is used when you configure dtmf disable.

Related command: timer dtmf.

Example

# Configure DTMF mode to receive and send R2 signaling.

[VG-cas0:0] dtmf enable

3.1.22 dtmf threshold (Digital Voice Subscriber-line)

Syntax

dtmf threshold { 0 | 1 }

undo dtmf threshold

View

E1 voice subscriber-line view

Parameter

0: Sets the DTMF code detection sensitivity to insensitive level.

1: Sets the DTMF code detection sensitivity to sensitive level.

For E1 voice subscriber lines, the default setting of DTMF code detection sensitivity is 0.

Description

Use the dtmf threshold command to configure the sensitivity of DTMF detection.

Use the undo dtmf threshold command to restore the default settings of DTMF detection sensitivity.

Example

# Set the DTMF code detection sensitivity for voice subscriber line 0:0 to 0.

[VG-voice-line0:0] dtmf threshold 0

3.1.23 effect-time

Syntax

effect-time number

undo effect-time

View

R2 CAS view

Parameter

number: Indicate the lower limit threshold of the debounce time of line signaling, with a value ranging from 10 to 40, and the unit being millisecond (ms). By default number is 40ms.

Description

Use the effect-time command to configure the debounce time of line signaling.

Use the undo effect-time command to restore the default value.

Only when the duration of line signaling exceeds this time threshold can the change of line be regarded as valid.

Related command: timeslots-set and pri-set.

Example

# Set the debounce time of line signaling as 20ms for the No.3 Timeslot set in the E1 subscriber-line 1.

[VG-cas0:3] effect-time 20

3.1.24 final-callednum

Syntax

final-callednum { enable | disable }

View

R2 CAS view

Parameter

enable: Enables the called number terminate signal to be sent back.

disable: Disables the called number terminate signal to be sent back.

Description

Use the final-callednum command to enable or disable the terminate signal to be sent to the terminating point after the called number is sent.

By default, the called number terminate signal is disabled.

In some countries, the R2 interregister signaling can be used to send the called number terminate signal after the called number is sent out, indicating that the called number transmission is completed. In this case, you can use this command to adjust the signaling exchange mode. When the terminating point receives the terminate signal, it stops requesting for the called number.

Related command: register-value digital-end.

Example

# Enable the called number terminate signal.

[VG-cas0:0] final-callednum enable

3.1.25 force-metering

Syntax

force-metering { enable | disable }

View

R2 CAS view

Parameter

enable: Enables the metering signal of R2 signaling.

disable: Disables the metering signal of R2 signaling.

Description

Use the force-metering command to enable or disable the metering signal of R2 signaling.

By default, the metering signal of R2 signaling is disabled.

If the opposite exchange supports the metering signal, H3C VG, as the terminating point, sends a forced-release signal instead of a clear-back signal when it terminates a call, so as to indicate that the called party has release the line and the call is terminated. And so metering signal collision can be avoided.

Example

# Enable the metering signal of R2 signaling.

[VG-cas0:0] force-metering enable

3.1.26 frame-format

Syntax

frame-format { crc4 | no-crc4 }

View

CE1/PRI interface view

Parameter

crc4: Specifies the frame format of the CE1/PRI interface to CRC4.

no-crc4: Specifies the frame format of the CE1/PRI interface to no-CRC4.

Description

Use the frame-format command to specify the frame format of the CE1/PRI interface.

Use the undo frame-format command to restore the default frame format of the interface.

By default, the frame format of the CE1/PRI interface is no-CRC4.

When the CE1/PRI interface operates in the CE1/PRI mode, both of the CRC4 and no-CRC4 can be used. CRC4 supports the 4-bit CRC of the physical frame while the no-CRC4 not.

Example

# Set the frame format of the interface E1 0 to CRC4.

[VG-E1-0] frame-format crc4

3.1.27 group-b

Syntax

group-b { enable | disable }

View

R2 CAS view

Parameter

enable: Enables signal exchange at Group-B stage of R2 signaling.

disable: Disables signal exchange at Group-B stage of R2 signaling, that is, backward Group-A signal A6 is used to complete register interoperation directly.

Description

Use the group-b command to enable or disable Group-B stage signal to complete register interoperation.

By default, Group-B stage signal is used to complete register interoperation, that is, the command is in enable state.

In some countries, R2 register interoperation does not support Group-B stage signal interoperation, or cannot correctly interpret Group-B signal value. Then you can use the group-b command to enable or disable Group-B signal interoperation.

Related command: register-value req-switch-groupb.

Example

# Enable Group-B signal to complete register interoperation.

[VG-cas0:0] group-b enable

3.1.28 line

Syntax

line e1-number: { ts-set-number | 15 }

undo line

View

POTS voice entity view

Parameter

e1-number: Indicate the serial number of E1 port to which this voice subscriber line belongs.

ts-set-number: Indicate the serial number of Timeslot set established successfully.

15 Indicate adopting the E1 voice ISDN PRI interface mode.

Description

Use the line command to configure the corresponding relationship between the POTS voice entity and the logic voice subscriber line.

Use the undo line command to cancel the corresponding relationship between the POTS peer and the logical voice subscriber line.

This command can be used in POTS voice entity view only and takes effect on E1 voice subscriber-line only.

After configuring the destination mode of voice entity by using the command match-template, it is required to use the command line to configure the corresponding relationship between the POTS voice entity and logic port, that is, to specify that via which port the routing should be performed toward this destination.

Use the command line e1-number:ts-set-number to select the voice subscriber line corresponding to Timeslot set in this E1 port as the routing output. Use the command line e1-number:15 corresponding to the ISDN PRI set in this E1 subscriber-line as the routing output.

Related command: timeslots-set, entity, and pri-set.

Example

# Configure the corresponding relationship between POTS voice entity 3 and No.1 Timeslot set in E1 port.

[VG-voice-dial-entity3] line 1:1

3.1.29 loopback

Syntax

loopback { local | remote }

View

CE1/PRI interface view

Parameter

local: Set the CE1/PRI interface to perform local loopback.

remote: Set the CE1/PRI interface to perform the remote loopback.

Description

Use the loopback command to set the CW1/PRI interface to perform the local and remote loopback.

Use the undo loopback command to disable the local and remote loopback.

By default, the local and remote loopback are both disabled.

The local loopback is used to test the interface and cable. During the normal operation, this is disabled.

Configure the link layer of the serial port formed by binding the CE1/PRI interface with timeslots to PPP, and set the remote loopback, it is normal that the link layer state is reported as DOWN.

Example

# Set the interface E1 0 to perform local loopback.

[VG-E1-0] loopback local

3.1.30 mfc (R2 CAS)

Syntax

mfc { block | open | query } timeslots timeslots-list

View

R2 CAS view

Parameter

block: Indicates blocking the MFC channel of the specified timeslot.

open: Indicates opening the MFC channel of the specified timeslot.

query: Indicates querying the MFC channel of the specified timeslot.

timeslots-list: Specifies a timeslot range. The expression includes the single digit, two digits separated by “,”, a pair of digits separated by “-”, or the combination form (e.g., 1-14, 15, 17-31). The value range of digits is the integer from 1 to 31.

Description

Use the mfc (R2) command to maintain of MFC channel of the specified timeslot.

To block the MFC channel means that this channel will no longer load the R2 interregister signaling information, that is, this channel is set manually as unavailable.

To open the MFC channel is the processes inverse to the blocking operation, which can re-set the channel as available and enable it to load R2 interregister signaling.

To query the MFC channel will display the busy/idle, opened/blocked status of channel in a real time way.

Related command: cas and ts.

Example

# Block the timeslots 1-15 in No.5 Timeslot set, and query the channel status of the timeslots 1-31.

[VG-cas0:5] mfc block timeslots 1-15

[VG-cas0:5] mfc query timeslots 1-31

3.1.31 mode

Syntax

mode zone-name { default-standard | custom }

mode custom

View

R2 CAS view

Parameter

zone-name: Name of the country or region. By default, it is set to the mode of ITU-T. It can be:

l argentina: Uses Argentinean R2 signaling standard.

l australia: Uses Australian R2 signaling standard.

l china: Uses Chinese R2 signaling standard.

l bengal: Uses Bengalee R2 signaling standard.

l brazil: Uses Brazilian R2 signaling standard.

l custom: Uses the R2 signaling mode defined by customer.

l hongkong: Uses Hongkong R2 signaling standard.

l india: Uses Indian R2 signaling standard.

l indonesia: Uses Indonesian R2 signaling standard.

l itu-t: Uses ITU-T R2 signaling standard.

l korea: Uses Korean R2 signaling standard.

l malaysia: Uses Malaysian R2 signaling standard.

l mexico: Uses Mexican R2 signaling standard.

l newzealand: Uses New Zealand R2 signaling standard.

l singapore: Uses Singaporean R2 signaling standard.

l thailand: Uses Thai R2 signaling standard.

default-standard: Initializes the related parameters of R2 signaling according to the mode in the current country, that is, initializes the values of the force-metering command.

custom: Notifies to use the customized parameters.

Description

Use the mode command to configure the R2 signaling mode in a country or region.

The implementation and parameters of R2 signaling vary in different countries. Therefore, the mode needs to be adjusted to enable H3C VG to exchange R2 signaling with the switching devices of different countries or regions. According to the configuration, the system automatically selects the appropriate subscriber-line state, service type, metering signal, and the signal values of C and D bits, etc. At present, it supports the modes in Brazil, Mexico, Argentina, India, New Zealand, Thailand, Bengal, South Korea, Hongkong, Indonesia, as well as the countries and regions that comply with ITU-T Recommendations.

In the custom mode, you can configure the specific signaling exchange process and signal values of R2 signaling, so as to adjust the R2 signaling in your country in a more flexible way.

If you change the parameters in a certain default mode, the system will automatically configure the mode zone-name custom command and displays the configuration command different from the default settings. If you configure the mode zone-name custom command and the current settings are the same as the default, the system will change to the corresponding default-standard mode.

Related command: register-value, force-metering, and effect-time.

Example

# Adopt the default standard of Hongkong R2 signaling mode.

[VG-cas0:0] mode hongkong default-standard

3.1.32 pri-set

Syntax

pri-set [ timeslot-list range ]

undo pri-set

View

CE1/PRI interface view

Parameter

range: Number of the bound timeslots, it is in the range 1 to 31. When specifying timeslots to be bound, you can configure a single timeslot using the form of number, or timeslots within a range using the form of number1-number2, or simultaneously specify multiple timeslots using the form of number1, number2-number3.

Description

Use the pri-set command to bind the timeslots of CE1/PRI interface to be a pri-set.

Use the undo pri-set command to cancel the existing binding.

By default, no pri-set is created.

In the case of implementing pri-set binding, timeslot 16 on the CE1/PRI interface will be used as D channels. Therefore, this timeslot cannot be bound independently. If only this timeslot is to be bound, the binding activity will fail.

In the case of binding the timeslots on the CE1/PRI interface to form a pri-set, timeslot 0 will be used to transmit synchronizing information, timeslot 16 as a D channel to transmit signaling, and the other timeslots as B channels to transmit data. At the time of binding, bind the timeslots except the timeslot 0 as one pri-set (as a D channel, the timeslot 16 will be automatically bound). The logic feature of this pri-set will be the same like that of an ISDN PRI interface. In the case of binding timeslots to be a pri-set, if timeslots to be bound are not specified, all the timeslots except timeslot 0 will be bound to form an ISDN PRI interface similar to 30B+D.

After the timeslot binding, the interface will automatically create a Serial interface owing the same logic feature as that of an ISDN PRI interface. The number of the Serial interface is serial number:15. number starts from the largest serial interface number plus 1.

Example

# Bind the timeslots 1, 2, 8 to 12 of the CE1/PRI interface as a pri-set.

[VG-E1-0] pri-set timeslot-list 1,2,8-12

3.1.33 re-answer

Syntax

re-answer { enable | disable }

View

R2 CAS view

Parameter

enable: Enables the originating point to support re-answer signal process.

disable: Disables the originating point to support re-answer signal process.

Description

Use the re-answer command to enable or disable the originating point to support re-answer signal process.

By default, the originating point does not support re-answer signal process.

In some countries, re-answer process is needed in R2 signaling. When the terminating point sends a clear-back signal, the originating point does not release the line right away, but maintains the call state instead. If it receives the re-answer signal from the terminating point in the specified time, it continues the call; otherwise, it disconnects the call after timeout.

Related command: answer and timer dl re-answer.

Example

# Enable the originating point to send re-answer signal.

[VG-cas0:0] re-answer enable

3.1.34 register-value

Syntax

register-value { billingcategory | callcreate-in-groupa | callingcategory | congestion | demand-refused | digit-end | nullnum | req-billingcategory | req-callingcategory | req-currentdigit | req-firstcallingnum | req-firstdigit | req-nextcallednum | req-nextcallingnum | req-lastfirstdigit | req-lastseconddigit | req-lastthirddigit | req-switch-groupb | subscriber-busy | subscriber-idle | subscriber-idle-nocharge | req-firstcallednum-groupc | req-currentcallednum-groupc | req-callednum-switchgroupa } value

register-value respond-req-callernum

undo register-value { all | billingcategory | callcreate-in-groupa | callingcategory | congestion | demand-refused | digit-end | nullnum | req-billingcategory | req-callingcategory | req-currentdigit | req-firstcallingnum | req-firstdigit | req-nextcallednum | req-nextcallingnum | req-lastfirstdigit | req-lastseconddigit | req-lastthirddigit | req-switch-bgroup | subscriber-busy | subscriber-idle | subscriber-idle-nocharge | respond-req-callernum | req-firstcallednum-groupc | req-currentcallednum-groupc | req-callednum-switchgroupa }

View

R2 CAS view

Parameter

billingcategory value: Specifies billing category value, which ranges from 1 to 16. By default, it is set to 2.

callcreate-in-groupa value: Specifies the directly-created calling signal, ranging from 1 to 16 with the default value 6.

callingcategory value: Specifies calling category value, which ranges from 1 to 16. By default, it is set to 1.

congestion value: Specifies congestion value, which ranges from 1 to 16. By default, it is set to 4.

demand-refused value: Specifies demand-refused value, which ranges from 1 to 16. By default, it is set to 12.

digit-end value: Specifies digit-end value, which ranges from 1 to 16. By default, it is set to 15.

nullnum value: Specifies null number value, which ranges from 1 to 16. By default, it is set to 5.

req-billingcategory value: Specifies request billing category value, which ranges from 1 to 16. By default, it is set to 5.

req-callingcategory value: Specifies request calling category value, which ranges from 1 to 16. By default, it is set to 3.

req-currentdigit value: Specifies request current digit value, which ranges from 1 to 16. By default, it is set to 16.

req-firstcallingnum value: Specifies request first calling number value, which ranges from 1 to 16. By default, it is set to 5.

req-firstdigit value: Specifies request first digit value, which ranges from 1 to 16. By default, it is set to 16.

req-nextcallednum value: Specifies request next called number value, which ranges from 1 to 16. By default, it is set to 1.

req-nextcallingnum value: Specifies request next calling number value, which ranges from 1 to 16. By default, it is set to 5.

req-lastfirstdigit value: Specifies request last first digit value, which ranges from 1 to 16. By default, it is set to 2.

req-lastseconddigit value: Specifies request last second digit value, which ranges from 1 to 16. By default, it is set to 7.

req-lastthirddigit value: Specifies request last third digit value, which ranges from 1 to 16. By default, it is set to 8.

req-switch-groupb value: Specifies request switch Group-B value, which ranges from 1 to 16. By default, it is set to 3.

subscriber-busy value: Specifies subscriber line busy value, which ranges from 1 to 16. By default, it is set to 3.

subscriber-idle value: Specifies subscriber line idle value, which ranges from 1 to 16. By default, it is set to 6.

subscriber-idle-nocharge value: Signal value of the subscriber-line idle (no-charge) state, ranging from 1 to 16 and the default value is 7.

respond-req-callernum: Specifies the terminating point whether to send the answer signal of the calling number. By default, no answer signal is sent.

req-firstcallednum-groupc value: Signal value of register in the Group C to request the first called number, ranging from 1 to 16 (default).

req-currentcallednum-groupc value: Signal value of register in the Group C to request the current called number, ranging from 1 to 16 (default).

req-callednum-switchgroupa value: Signal value of register requesting the next called number and forwarding the state of the Group A. It ranges from 1 to 16 and defaults to 1.

all: Deletes all register signal values.

Description

Use the register-value command to configure the value of all register signals of R2 signaling.

Use the undo register-value command to restore the default value. Signal value 16 means the corresponding signal function does not exist. For example, if the request last first digit function does not exist in some countries, the value of req-lastfirstdigit is 16.

By configuring the register-value command, you can send the specified request signal and require the opposite exchange to send back the corresponding answer signal. For example, using the register-value callingcategory command, you can enable the terminating point to send the specified signal to require the originating point to send back the calling category.

The register-value billingcategory command is used to configure KA signal of R2 signaling. That is, the originating point sends a calling category signal to the originating toll office or originating international exchange, which provides two kinds of information: billing category of this connection (paid at the specified time or at once, or toll free) and subscriber level (with or without priority).

The register-value callingcategory command is used to configure KD signal of R2 signaling, i.e., calling category. It functions to identify whether break-in and forced- release can be implemented by or on the calling party.

The register-value subscriber-idle command is used to configure KB signal of R2 signaling. It indicates subscriber status (such as idle), and acknowledges and controls connection. You should make sure that KB values of the both ends are the same. H3C VGs or VGs are used at both ends, you should make sure that KB values of the both ends are the same. If PBX is used at one end, and a VG is used at the other end, you should adjust the KB value of the VG to keep it consistent with that of the PBX.

& Note:

In some countries, the register signal encoding scheme does not support all the register signals described above. For example, there is no billingcategory but callingcategory in ITU-T Recommendations. Therefore, you should use the default values without modifying the configuration, unless there are special requirements.

Related command: group-b { enable/disable }.

Example

# Request the originating point to send calling category by configuring a backward signal (signal value 7).

[VG-cas0:0] register-value req-callingcategory 7

3.1.35 renew

Syntax

renew A-bit B-bit C-bit D-bit

undo renew

View

R2 CAS view

Parameter

A-bit, B-bit, C-bit, and D-bit: Indicate the default of each signal bit in transmission, with the value being 0 or 1. By default A-bit B-bit C-bit D-bit is 1 1 1 1.

Description

Use the renew command to configure the signal values of C bit and D bit.

Use the undo renew command to restore the default value.

In the R2 signaling A-bit and B-bit are used to transmit the valid information, and the actual transmission signal has nothing to do with the setting value. C-bit and D-bit do not transmit the valid information, and generally the set signal value is adopted as the transmission signals. Therefore, for the R2 signaling, this command only makes sense to C-bit and D-bit.

Use this command to adjust the values of C and D bits according to the line signaling encoding standards in different countries. For example, in China, the values of C and D bits of R2 signaling are fixed: 1, 1. However, in most other countries, the values of C and D bits can only be 0, 1.

Related commands: cas, reverse.

& Note:

This command is available only in the mode zone-name custom mode, but not in the mode custom mode. Namely, the mode custom command does not set the c and d bits.

Example

# Configure the signal values of both the C-bit and D-bit of R2 line signaling as 1.

[VG-cas0:5] renew 0 0 1 1

3.1.36 reset voice r2

Syntax

reset voice r2

View

Any view

Parameter

None

Description

Use the reset voice r2 command to reset the call statistics of R2 signaling.

Related command: display voice r2.

Example

# Reset the call statistics of R2 signaling.

[VG] reset voice r2

3.1.37 respond-reqcallernum

Syntax

respond-reqcallernum

undo respond-reqcallernum

View

R2 CAS view

Parameter

None

Description

Use the respond-reqcallernum command to request the originating VG to send caller number response signals.

Use the undo respond-reqcallernum command to cancel the configuration of requesting the originating VG to send caller number response signal.

By default, the originating VG is not request to send response signals.

In some countries, when the terminating VG sends the signal to request for the caller number, the originating VG is request to answer whether it supports sending caller number response signals. Use the command to determine whether the originating VG is requested to send response signals, making it compatible with these modes.

Example

# Configure to require the originating VG to send caller number response signals.

[VG-cas0:0] respond-reqcallernum

3.1.38 reverse

Syntax

reverse A-bit B-bit C-bit D-bit

undo reverse

View

R2 CAS view

Parameter

A-bit, B-bit, C-bit and D-bit Indicate whether to perform the inversion of each signal bit, with the value of each bit being 0 or 1. By default, A-bit B-bit C-bit D-bit is 0 0 0 0, that is, the function of inversion change is disabled.

Description

Use the reverse command to configure the inversion mode of line signals.

Use the undo reverse command to restore the default value.

This command can be used to perform the inversion change to A, B, C, and D bits prior to sending and after receiving the line signal, that is, 0 is changed to 1, and 1 to 0. If the value of one bit is 1, it indicates this bit is needed to invert.

Related command: cas and renew.

Example

# Configure that the B bit and D bit of R2 line signaling require the inversion.

[VG-cas0:0] reverse 0 1 0 1

3.1.39 seizure-ack

Syntax

seizure-ack { enable | disable }

View

R2 CAS view

Parameter

enable: Enables the originating point to request the terminating point to send seizure acknowledgement signal.

disable: Disables the originating point from requesting the terminating point to send seizure acknowledgement signal.

Description

Use the seizure-ack command to specify the terminating point whether to send seizure acknowledgement signal.

By default, the originating point requests the terminating point to send seizure acknowledgement signal.

Normally, the terminating point sends seizure acknowledgement signal when it receives a seizure signal from the originating point. However, in some countries R2 allows the terminating point not to send seizure acknowledgement signal. In this case, you can use the seizure-ack command to enable adaptation to such a signaling encoding scheme. If the originating point does not request the terminating point to send the seizure acknowledge signal, the terminating point needs not respond to the originating point with a seizure acknowledge signal when it receives a seizure signal, and can continue the calling process. If the originating point is configured to request the terminating point to send the seizure acknowledge signal, the terminating point must reply with a seizure acknowledge signal; otherwise the call will be disconnect.

Related command: timer dl seizure.

Example

# Disable the originating point from requesting the terminating point to send seizure acknowledgement signal.

[VG-cas0:0] seizure-ack disable

3.1.40 select-mode

Syntax

select-mode [ max | maxpoll | min | minpoll ]

View

R2 CAS view, digital E&M signaling view

Parameter

max: Indicate the maximum selection.

maxpoll: Indicate the maximum polling selection.

min: Indicate the minimum selection.

minpoll: Indicate the minimum polling selection.

Description

Use the select-mode command to configure the E1 trunk selection mode.

By default, the trunk selection mode is min.

The proper selection policy can not only enables each timeslot in E1 trunk to have the balanced opportunity to be used but also helps to enhance the speed of selecting idle timeslot, so as to improve the telephone connection speed.

The parameter max indicates the maximum selection, selecting the timeslot of the maximum serial number from the currently available timeslots.

The parameter maxpoll indicates the maximum polling selection. When used for the first time, select the timeslot of the maximum serial number from the currently available timeslots, and next time select the available timeslot whose serial number is less than it. For example, No.31 and No.29 timeslots are unavailable in 32 timeslots, so select firstly No.30 timeslot, and select No.28 timeslot secondly.

The parameter min indicates the minimum selection, selecting the timeslot of the minimum serial number from the currently available timeslots.

The parameter minpoll indicates the minimum polling selection. When used for the first time, select the timeslot of the minimum serial number from the currently available timeslots , and next time select the available timeslot whose serial number is bigger than it. For example, No.1 and No.3 timeslots are unavailable in 32 timeslots, so select firstly No.2 timeslot, and select No.4 timeslot secondly.

Use the command select-mode without any parameter can restore the default trunk timeslot selection mode.

Related command: cas and trunk-direction.

Example

# Configure the trunk selection mode as max for the No.5 Timeslot set in No.0 E1 subscriber-line.

[VG-cas0:5] select-mode max

3.1.41 sendring

Syntax

sendring { ringback | ringbusy } { enable | disable }

View

R2 CAS view

Parameter

ringback { enable | disable }: Enables/disables the sending of ringback signal.

ringbusy { enable | disable }: Enables/disables the sending of ringbusy signal.

Description

Use the sendring command to enable or disable the terminating point to send ringback tone or busy tone signal to the calling party.

By default, the busy tone signal, not the ringback tone signal, is sent.

In some regions a PBX, as the originating point, might not send ring-back tone to the calling party during call connecting process. To avoid call connection failure due to the calling party not able to hear the corresponding tone, you can manually configure the sendring command. If the VG works as the terminating point, it sends the corresponding tone to the originating point according to call connection situations.

Related command: timer ring.

Example

# When time slot 5 on the E1 port 0 works as the terminating point, enable it to send ring-back tone to the originating point.

[VG-cas0:5] sendring ringback enable

3.1.42 signal-value

Syntax

signal-value { received idle | received seize | transmit-bits idle | transmit seize } A-bit B-bit C-bit D-bit

undo signal-value { received idle | received seize | transmit-bits idle | transmit seize }

View

Digital E&M voice subscriber-line view

Parameter

received idle: The digital E&M subscriber-line receives the idle signaling.

received seize: The digital E&M subscriber-line receives the seizure signaling.

transmit idle: That the digital E&M subscriber-line transmits the idle signaling.

transmit seize: The digital E&M subscriber-line transmits the seizure signaling.

A-bit, B-bit, C-bit, D-bit: The default value of each signaling bit in a transmission. It is valued to be either 0 or 1. By default, the ABCD bits value of the received and transmitted idle signaling on the digital E&M subscriber-line are 1 1 0 1. The ABCD bits value of the received and transmitted seizure signaling on the digital E&M subscriber line are 0101.

Description

Use the signal-value command to configure the digital E&M subscriber-line to receive and transmit the ABCD bits of idle signaling and seizure signaling.

Use the undo signal-value command to restore the bits of the corresponding signaling to the default value.

When the VG and its peer device, such as a PBX, communicate using the digital E&M signaling, it should be ensured that they interpret the ABCD bits of the received and transmitted idle signaling and seizure signaling in the same way. That is, the signaling of the same type should have the same bit value at both ends.

Related command: subscriber-line.

Example

# Set the ABCD bits of seizure signaling transmitted by the digital E&M subscriber-line are 1 0 1 1.

[VG-voice-line0:0] signal-value transmit seize 1011

3.1.43 special-character

Syntax

special-character character value number

View

R2 CAS view

Parameter

character: A special character, which can be #, *, A, B, C, or D.

value number: Code of the register signal, ranging from 11 to 15.

Description

Use the special-character command to configure the supported special characters during register signal exchange.

By default, no special character is configured.

In some countries, besides numerical information, register forward Group I signal of R2 signaling also supports the information containing special characters, such as # and *. To encode these special characters, you need to use the special-character command.

& Note:

It is not allowed to use the special-character command to configure one special character to different signal codes.

Example

# Configure register signal code of # to 11.

[VG-cas0:0] special-character # value 11

3.1.44 subscriber-line

Syntax

subscriber-line e1-number: { ts-set-number | 15 }

View

Voice view

Parameter

e1-number: Indicate the voice subscriber line number generated in creating Timeslot set or ISDN PRI set.

ts-set-number: Indicate the number of Timeslot set created successfully.

15: Indicate the subscriber line is generated in creating the ISDN PRI set on E1 subscriber line.

Description

Use the subscriber-line command to enter the E1 voice subscriber-line view.

After creating the Timeslot set successfully, the system will generate the subscriber line corresponding to this Timeslot set according to current E1 subscriber-line number and Timeslot set number, and the subscriber line number is “E1 subscriber-line number: Timeslot set number”.

After configuring the ISDN PRI set successfully, the system will generate the subscriber line corresponding to this PRI set according to the number of E1 subscriber-line where the current PRI subscriber-line is located, and the subscriber line number is “E1 subscriber-line number: 15”.

Related command: timeslots-set and pri-set.

Example

# Enter the view of subscriber line 0:5.

[VG-voice] subscriber-line 0:5

[VG-voice-line0:5]

3.1.45 timer (digital E&M)

Syntax

timer dial-interval seconds

timer { ring-back | wait-digit } { seconds | infinity }

undo timer { dial-interval | ring-back | wait-digit }

View

Digital E&M voice subscriber-line view

Parameter

dial-interval seconds: Sets the max waiting time in seconds between two digit numbers. The value range is 1 to 300, the default value is 4 seconds.

ring-back seconds: Sets the timeout time in seconds for the calling party to wait for the ringback response from the called party. The value range is 5 to 60000, and the default value is 60 seconds.

wait-digits second: Sets the timeout time in seconds for the called party to wait for the called number. The value range is 3 to 600, and the default value is 5 seconds.

infinity: Indicates that there is no time limit, that is, timeout will not occur.

Description

Use the timer command to configure the timeout values of the signals in the digital E&M signaling.

Use undo timer command to restore the timeout values to the default value.

The timer ring-back seconds command can be used at the digital E&M subscriber-line to set the timeout time for a calling party to wait for the ringback response from the called party.

The timer wait-digit seconds command can be used at the digital E&M subscriber-line to set the timeout time for a called party to wait for the called number.

Related command: timing.

Example

# Configure the timeout it takes to wait for the ringback response at subscriber-line 3 to 30 seconds.

[VG-voice-line0:3] timer ring-back 30

3.1.46 timer dl (R2)

Syntax

timer dl { answer | clear-back | clear-forward | seize | re-answer | release-guard } time

undo timer dl { answer | clear-back | clear-forward | seize | re-answer | release-guard }

View

R2 CAS view

Parameter

answer time: Timeout time in which R2 waits for an answer signal. The terminating point should send back an answer signal in the specified time after a seizure acknowledgement signal is sent. It ranges from 100 to 120,000ms. By default, it is set to 60,000ms.

clear-back time: Timeout time of R2 clear-back signal. After it sends a clear-back signal, the terminating point should recognize the forward signal sent by the originating point in the specified time. It ranges from 100 to 60,000ms. By default, it is set to 1,000ms.

clear-forward time: Timeout value of R2 clear-forward signal. After the originating point sends a clear-forward signal, the terminating point should send back a corresponding line signal in the specified time, such as clear-back signal or release guard signal. It ranges from 100 to 60,000ms. By default, it is set to 1,000ms.

seize time: Timeout value of R2 seizure signal. After the originating point sends a seizure signal, the terminating point should send back a seizure acknowledgement signal in the specified time. It ranges from 100 to 5,000ms. By default, it is set to 1,000ms.

re-answer time: Timeout value of R2 re-answer signal. When the originating point recognizes the clear-back signal, if the terminating point does not send a re-answer signal in the specified time, the originating point releases the line. It ranges from 100 to 60,000ms. By default, it is set to 1,000ms.

release-guard time: Timeout value of R2 release guard signal. When the originating point sends a clear-forward signal, the terminating point should send a release guard signal in the specified time after it sends back a clear-back signal. It ranges from 100 to 60,000ms. By default, it is set to 1,000ms.

Description

Use the timer dl command to configure the timeout value of line signals of R2 signaling.

Use the undo timer dl command to restore the default value.

Example

# Configure the timeout value of seizure signal of R2 signaling to 300ms.

[VG-cas0:0] timer dl seize 300

3.1.47 timer dtmf (R2)

Syntax

timer dtmf time

undo timer dtmf

View

R2 CAS view

Parameter

time: Time interval for R2 signaling to send DTMF signals, ranging from 0 to 10000 milliseconds. By default, it is set to 0 milliseconds.

Description

Use the timer dtmf command to configure the time interval for sending DTMF signals.

Use the undo timer dtmf command to restore the default value.

Normally, the originating point sends a DTMF signal upon receiving the line seizure acknowledgement signal. Using this command, you can configure the device at the originating point to send a DTMF signal after the specified time interval, so as to make it appropriate to the number receiving process of the peer PBX,

Related command: dtmf { enable/disable }.

Example

# Configure the R2 signaling to send a DTMF signal 800 milliseconds after the reception of the seizure acknowledgement signal.

[VG-cas0:0] timer dtmf 800

3.1.48 timer register-complete (R2)

Syntax

timer register-complete { group-b } time

undo timer register-complete { group-b }

View

R2 CAS view

Parameter

group-b time: Timeout value in which the originating point waits for Group-B signal of R2 signaling. The terminating point should send Group-B signal within the specified time when switching to Group-B exchange. It ranges from 100 to 90,000ms. By default, it is set to 30,000ms.

Description

Use the timer register-complete command to configure the timeout value for register signals of R2 signaling.

Use the undo timer register-complete command to restore the default value.

Related command: timer dl.

Example

# Configure the maximum Group-B signal exchange time to 10,000 ms during connecting process.

[VG-cas0:0] timer register-complete group-b 10000

3.1.49 timer register-pulse (R2)

Syntax

timer register-pulse persistence time

undo timer register-pulse

View

R2 CAS view

Parameter

persistence time: Duration time of the register pulse signal of R2 signaling, ranging from 100 to 1,000 milliseconds. By default, it is set to 150±30 milliseconds.

Description

Use the timer register-pulse command to configure the persistence time of the register pulse signal of R2 signaling (A3, A4 and A6, etc.).

Use the undo timer register-pulse command to restore the default value.

When the terminating point sends a backward register pulse signal, such as A3, the signal must persist for a specified time range. When the originating point receives the pulsed A3 signal, it has to send a forward Group II signal. When the originating point recognizes the pulse signal A4, A6, or A15, it stops sending any forward signal, and terminates the register signal exchange.

Related command: timer register-complete.

Example

# Set the persistence time of the register pulse signal of R2 signaling to 300ms.

[VG-cas0:0] timer register-pulse persistence 300

3.1.50 timer ring (R2)

Syntax

timer ring { ringback | ringbusy } time

undo timer ring

View

R2 CAS view

Parameter

ringback time: Timeout interval before ring-back tone is sent, ranging from 1,000 to 90,000ms. By default, it is set to 60,000ms.

ringbusy value: Timeout interval before busy tone is sent, ranging from 1,000 to 90,000ms. By default, it is set to 30,000ms.

Description

Use the timer ring command to configure the maximum time range before the signal tone of R2 signaling is sent.

Use the undo timer ring command to restore the default value.

After manually configuring the sending command, you can enable the device at the originating point to send ring-back and busy tones to the calling party. The timer ringing command helps you identify these signals.

Related command: sendring.

Example

# Configure the timeout time for ring-back tone of R2 signaling to 10,000ms.

[VG-cas0:0] timer ring ringback 10000

3.1.51 timeslot-set

Syntax

timeslot-set ts-set-number timeslots-list signal { e&m-delay | e&m-immediate | e&m-wink | r2 }

undo timeslot-set ts-set-number

View

CE1/PRI interface view

Parameter

ts-set-number: Specify the identification number of a certain Timeslot set, with the value range being the integer from 0 to 30.

timeslots-list: Specify a timeslot range. The expression includes the single digit, two digits separated by “,”, a pair of digits separated by “-”, or the combination form (e.g., 1-14, 15, 17-31). The value range of digits is the integer from 1 to 31.

signal signal Specify the binding signaling mode of this Timeslot set, which is generally used to configure the signaling mode adopted by the central office. It includes the following types of signaling:

l e&m-delay Adopt the delay-start mode in the digital E&M signaling.

l e&m-immediate Adopt the immediate-start mode in the digital E&M signaling.

l e&m-wink Adopt the wink-start mode in the digital E&M signaling.

l r2 Specify the signaling mode of adopting ITU-T Q.421 digital line signaling R2, which is the most common configuration signaling.

Description

Use the timeslot-set command to configure the timeslot set to perform R2 signaling and digital E&M signaling configurations.

Use the undo timeslot-set command to cancel the specified timeslot set.

By default, no TS set is configured, and if the digital E&M signaling is used, then adopts e&m-immediate mode.

The Timeslot set is actually the logical voice subscriber line abstracted from the physical E1 interface, mainly used for the configuration of R2 signaling, digital E&M signaling and other voice functions. In one E1 interface only one Timeslot set can be defined. In defining the timeslot range of Timeslot set, the timeslot range can be distributed is 1-15 and 17-31, and the No.16 timeslot is reserved as the transmission channel of the out-of-band signaling.

Given the digital E&M signaling delay-start mode (e&m-delay) is adopted. If the calling side off-hooks to occupy the trunk line, the connected peer (e.g., PBX) will also enter the off-hook state to answer the calling party and will remain in that state until it is ready for receiving the address message. In this case, the PBX enters the hook-up state (this interval is the delay-dial period). The calling party sends the address message, and PBX connects this call to the called party, and thus the two parties can begin their communications.

When the digital E&M signaling e&m-immediate mode is adopted, the calling party will off-hook to wait for the confirmation of time, and after that, it will send the dialing address message to the connected peer such as a PBX. During the process, it will not detect whether the PBX is ready to receive.

When the digital E&M signaling e&m-wink mode is adopted, the calling party will first off-hook to occupy the trunk line while the connected peer (e.g., a PBX) remains in the hook-up state until it receives the connecting signal from the calling party. In this case, the wink signal sent by the PBX indicates that it has been ready. Upon receiving the wink signal, the calling party begins to send the address message and the PBX will connect the call to the called party and thus the two parties can begin their communications.

Similarly, the receive end can end the transfer of backward signal only after detecting that the transfer of forward signal has been stopped. The advantage of this mode is the high reliability of signal transfer, but the transfer speed is relatively slow over the lines with the long transfer time delay.

Only after establishing TS set successfully can the command subscriber-line be used to enter the subscriber line and configure the voice-related attributes.

Related command: subscriber-line and cas.

Example

# Establish the Timeslot set, with its number being 5, timeslot including 1-31, signaling being the R2 signaling.

[VG-E1-0] timeslot-set 5 1-31 signal r2

3.1.52 trunk-direction

Syntax

trunk-direction timeslots timeslots-list { in | out | dual }

undo trunk-direction timeslots timeslots-list

View

R2 CAS view

Parameter

timeslots-list: Specify the range of trunk timeslot. The expression includes the single digit, two digits separated by “,”, a pair of digits separated by “-”, or the combination form (e.g., 1-14, 15, 17-31). The value range of digits is the integer from 1 to 31.

in: Indicate the trunk is the incoming trunk.

out: Indicate the trunk is the outgoing trunk.

dual: Indicate the trunk is the bidirectional trunk.

Description

Use the trunk-direction command to configure the E1 trunk direction.

Use the undo trunk-direction command to restore the default value.

By default, configure the bidirectional trunk.

When configuring the E1 trunk direction as incoming trunk, this trunk will not load any outgoing call. When configuring the E1 trunk direction as outgoing trunk, this trunk can only be used for the outgoing call and not for incoming call. When configuring it as bidirectional trunk, it can load the outgoing call and incoming call respectively according to the initiative of originating call.

To normalize the E1 communication, if the E1 trunk adopts the mode of incoming trunk or outgoing trunk, then one end of it must be ensured as incoming and other as outgoing, or the connection will fail. If both ends of E1 trunk adopt the bidirection mode, it is required to use the command of select-mode to adjust the trunk selection policy to avoid the simultaneous hold of timeslot by the two sides of communication.

In configuration avoid that one end is bidirectional trunk while the other end is outgoing trunk, or the call from the end configured as the bidirectional trunk will always fail.

Related command: cas and select-mode.

Example

# Configure the trunk direction as bidirectional trunk for No.5 Timeslot set in No.0 E1 subscriber-line.

[VG-cas0:5] trunk-direction timeslots 1-31 dual

3.1.53 ts

Syntax

ts { block | open | query | reset } timeslots timeslots-list

View

R2 signaling and digital E&M signaling views

Parameter

block: Indicate blocking the trunk circuit of the specified timeslot.

open: Indicate opening the trunk circuit of the specified timeslot.

query: Indicate querying the trunk circuit of the specified timeslot.

reset: Indicate resetting the trunk circuit of the specified timeslot.

Description

Use the ts command to maintain the trunk circuit of the specified timeslot. Only the query function is available in digital E&M signaling view.

To block the trunk circuit means that the circuit no longer loads the service information, that is, this circuit is set manually as unavailable.

To open the trunk circuit is the inverse process of the blocking operation, which can reset the trunk circuit as available and enable it to load service information.

To query the trunk circuit will display the busy/idle, opened/blocked status of the circuit in a real time way.

To reset the trunk circuit refers to re-initializing the state of trunk circuit. Generally, if the circuit state cannot be restored to normal in blocking or opening the circuit manually, it is required to perform the resetting. If the circuit cannot be reset automatically and correctly because of other reasons, it is generally required to reset manually the circuit, too.

Related command: cas and mfc.

Example

# Reset the circuit of timeslots 1-15 in No.5 Timeslot set and query the status of the circuit of timeslots 1-31.

[VG-cas0:5] ts reset timeslots 1-15

[VG-cas0:5] ts query timeslots 1-31

3.1.54 update

Syntax

update slot slot-number ftpserver { ip-address | host-name } filename filename [ username name | password password | port port ]*

View

System view

Parameter

slot slot-number: Slot number of the card.

ftpserver { ip-address | host-name }: The IP address or host name of the FTP server where the update file is located.

filename filename: Name of the on-card program update file, ranging 1 to 63 characters.

username name: Legal user name on the FTP file server, ranging 1 to 21 characters.

password password: Legal user password on the FTP file server, ranging 1 to 21 characters.

port port: The serving port of the FTP file server.

Description

Use the update command to live-update the functional program of the specified E1 voice interface card.

Example

# Update the functional program of the specified E1 voice interface card on slot 0.

[VG] update slot 0 ftpserver switch filename E1viram.bin username h3c password 123456

Chapter 4 ISDN Configuration Commands

4.1 ISDN Configuration Commands

4.1.1 debugging isdn

Syntax

debugging isdn { cc | q921 | q931 | qsig { alarm | all | call-state | error | information | message } } [ interface type number ]

View

Any view

Parameter

cc: Enable the ISDN CC packets debugging.

q921: Enable the ISDN Q921 packets debugging.

q931: Enable the ISDN Q931 packets debugging.

qsig: Enable the ISDN Qsig packets debugging.

alarm: To display alarming information.

all: To display all debugging information.

call-state: To display calling status and timeout information of timer.

error: To display error information.

information: To display general prompt information.

message: To display description information of message.

interface type number: Interface type and number. This parameter is used for debugging ISDN signaling on a specified ISDN interface. If no interface is specified by this parameter, ISDN signaling debugging on all the ISDN interfaces will be enabled.

Description

Use the debugging isdn command to enable ISDN signaling debugging.

Example

# Enable the ISDN Q931 packets debugging.

[VG] debugging isdn q931 message

4.1.2 display isdn active-channel

Syntax

display isdn active-channel [ interface type number ]

View

Any view.

Parameter

interface type number: Interface type and number. If no interface is specified, all the current call information will be displayed.

Description

Use the display isdn active-channel command to view the current call information on the ISDN interface.

The output of the command can help you troubleshoot for the ISDN calls.

Example

# Display the active call on the Serial 0:15.

[VG] display isdn active-channel interface Serial0:15

Serial0:15:

-------------------------------------------------------------

Channel Call Call Calling Calling Called Called

Info Property Type Number Subaddress Number Subaddress

B1 Digital Out 8810124

B2 Analog In 8810118 380 8810150 2201

-------------------------------------------------------------

Table 4-1 Description on the fields of the display isdn active-channel command

Item	Description
Channel Info	Name of the call channel
Call Property	Call Property. It may be Digital of Analog.
Call Type	Call Type. It may be Out of In
Calling Number	Calling Number
Calling Subaddress	Calling Subaddress
Called Number	Called Number
Called Subaddress	Called Subaddress

4.1.3 display isdn call-info

Syntax

display isdn call-info [ interface type number ]

View

Any view.

Parameter

interface type number: Interface type and number. If no interface is specified, all the current status on the ISDN interface will be displayed.

Description

Use the display isdn call-info command to view the current status of the ISDN interface.

The information output by this command includes the states of the layers of the ISDN protocol on the interface, including Q.921, Q.931, QSIG and CC. Using this command, users can make fault diagnosis. If no interface is specified, the current states of all ISDN interfaces are displayed.

Related command: display interfaces.

Example

# Display the current status of the ISDN interface.

[VG] display isdn call-info

Bri0(dss1):

Physical Layer: ACTIVE

Link Layer 1: TEI = 65, State = MULTIPLE_FRAME_ESTABLISHED

Link Layer 2: TEI = NONE, State = TEI_UNASSIGNED

Link Layer 3: TEI = NONE, State = TEI_UNASSIGNED

Link Layer 4: TEI = NONE, State = TEI_UNASSIGNED

Link Layer 5: TEI = NONE, State = TEI_UNASSIGNED

Link Layer 6: TEI = NONE, State = TEI_UNASSIGNED

Link Layer 7: TEI = NONE, State = TEI_UNASSIGNED

Network Layer: 1 connection(s)

Connection 1:

CCIndex: 0x00ae, State: Active, CES: 1, Channel: 0x00000002

Call Type: Out

TEI: 65

Calling_Num [:Sub]: 006

Called_Num [:Sub]: 007

Table 4-2 Description on the fields of the display isdn call-info command

Field	Description
Bri0(dss1): Physical Layer: ACTIVE	ISDN DSS1 protocol is applied on the Bri0 interface and the D channel is active.
Link Layer 1: TEI = 65, State = MULTIPLE_FRAME_ESTABLISHED	The related parameters of the first Q921 protocol on the ISDN link layer. The TEI value is 65, the state of the Q921 is MULTIPLE_FRAME_ESTABLISHED.
Network Layer: 1 connection(s)	The interface connects to one network layer.
CCIndex:0x0003	Call index is 0x0003.
State: Active	The call state is active.
CES:1	The CES of the Q921 applied in the call is 1.
Channel:0x00000002	Channel mapping (timeslot map): No. 2 B-channel of Bri
TEI: 65	The TEI value for the call is 65.
Calling_Num[:Sub]:006	The calling number is 006.
Called_Num[:Sub]:007	The called number is 007.

4.1.4 display isdn dss1-parameters

Syntax

display isdn dss1-parameters

View

Any view.

Parameter

None

Description

Use the display isdn dss1-parameters command to view the value of the ISDN timer.

This command is mainly intended for technical support engineers in their fault diagnosis and troubleshooting.

Related command: display interfaces.

Example

# Display the value of the isdn timer.

[VG] display isdn dss1-parameters

ISDN Q921 global parameters:

T200(sec) T202(sec) T203(sec) N200 K(Bri) K(Pri)

1 2 10 3 1 7

ISDN Q931 global timers:

Timer-Number Value(sec)

T301 240

T302 15

T303 4

T304 30

T305 30

T308 4

T309 90

T310 40

T313 4

T314 4

T316 120

T317 10

T318 4

T319 4

T321 30

T322 4

Table 4-3 Description on the fields of the display isdn dss1-parameters command

Item	Description
T200(sec)	Timer value of retranslation of ISDN Q921 protocol, in seconds.
T202(sec)	Timer value of retranslation of TEI requiring message of ISDN Q921 protocol, in seconds.
T203(sec)	The maximum amount of link idle time of ISDN Q.921 protocol, in seconds.
N200	The Max. times of retranslation.
K(Bri)	The permitted maximum number of unacknowledged frames on ISDN BRI port.
K(Pri)	The permitted maximum number of unacknowledged frames on ISDN PRI port.
Timer-Number	Timer number of ISDN Q931.
Value(sec)	Timer value of the ISDN Q931, in seconds.

4.1.5 display isdn q931-timer

Command

display isdn q931-timer [ interface type num ]

View

Any view

Parameter

interface type num: Display the timer information of a specified interface.

Description

Use the display isdn q931-timer command to view the Q931 timer information.

Example

# Display the Q931 timer information on Serilal0:15.

[VG] display isdn q931-timer interface Serial0:15

Serial0:15

Timer-Number Current Value(sec) Default Value(sec)

T301 180 180

T302 15 15

T303 5 5

T304 30 30

T305 30 30

T308 5 5

T309 90 90

T310 30 30

T313 5 5

T314 4 4

T316 120 120

T317 10 10

T318 4 4

T319 4 4

T321 30 30

T322 4 4

T323 4 4

T324 4 4

T325 30 30

4.1.6 display isdn qsig-timer

Syntax

display isdn qsig-timer [ interface type number ]

View

Any view.

Parameter

interface type number: Interface type and number. This parameter is used to display the values of ISDN QSIG signaling timers on a particular ISDN interface. If no interface is specified by this parameter, the values of timers on all the interfaces will be displayed.

Description

Use the display isdn qsig-timer command to view the value of all of the ISDN QSIG signaling timers.

This command is mainly intended for technical support engineers in their fault diagnosis and troubleshooting.

Related command: isdn qsig-timer, display isdn call-info.

Example

# Display the values of the ISDN QSIG signaling timers on the interface Serial 0:15.

[VG] display isdn qsig-timer interface serial 0:15

Serial0:15

Timer-Number Current Value(sec) Default Value(sec)

T301 180 180

T302 15 15

T303 5 5

T304 30 30

T305 30 30

T308 5 5

T309 90 90

T310 30 30

T313 5 5

T314 4 4

T316 120 120

T317 10 10

T318 4 4

T319 4 4

T321 30 30

T322 4 4

T323 4 4

T324 4 4

T325 30 30

Table 4-4 Description on the fields of the display isdn qsig-timer command

Item	Description
Timer-Number	Timer number of ISDN QSIG
Current Value (sec)	Current timer value (in seconds) of the ISDN QSIG
Default Value (sec)	Default timer value (in seconds) of the ISDN QSIG

4.1.7 isdn callingnum

Syntax

isdn callingnum calling-number

undo isdn callingnum

View

cE1/PRI interface view

Parameter

calling-number: Calling number.

Description

Use the isdn callingnum command to have the messages from a calling party to a called party carry the calling number.

Use the undo isdn callingnum command to delete calling number in the messages from a calling party.

By default, the calling number information is carried by the ISDN calling message.

If the calling number information is carried in the message when the calling party initiates a call, the destination VG will send the calling information to the called party; if the calling number information is not carried in the message when the calling party initiates a call, the destination VG will send the number configured by this command to the called party as the calling number.

If a calling party has configured this command on its ISDN interface, the called party will be able to see the calling number by viewing the call history information.

Related command: isdn ignore callednum.

Example

# Configure interface Serial 0:15 so that the message from a calling party to a called party carries the calling number.

[VG-Serial0:15] isdn callingnum 8060170

4.1.8 isdn check-called-number

Syntax

isdn check-called-number check-index called-party-number [ :sub-address ]

undo isdn check-called-number check-index

View

cE1/PRI Interface view

Parameter

check-index: Index of the called number or sub-address, in the range of 1 to 3.

called-party-number: Called number which is a character string of 1 to 20 numerical characters.

sub-address: Case insensitive sub-address which is a character string of 1 to 20 numerical or English alphabetic characters.

Description

Use the isdn check-called-number command to configure the VG to check the called numbers and sub-addresses in the received calls.

Use the undo isdn check-called-number command to cancel the existing setting.

By default, the VG does not check the called number and the sub-address.

This command is used to set checked items in the incoming call. When the sub-address is set, this call will be rejected in either case that the peer does not send the sub-address or a wrong one is sent.

If you want to check several groups of called number and sub-address, you can use this command to set at most 3 groups of called numbers and sub-addresses to be checked.

Example

# Set the called number to be checked for interface Serial 0:15 incoming call to 66668888.

[VG-Serial0:15] isdn check-called-number 1 66668888:23

4.1.9 isdn communicate italy

Syntax

isdn communicate italy

undo isdn communicate italy

View

cE1/PRI interface view

Parameter

None

Description

Use the isdn communicate italy command to enable the Telecom Italia ISDN configuration.

Use the undo isdn communicate italy command to disable the Telecom Italia ISDN configuration.

By default, the Telecom Italia ISDN configuration is disabled.

In case of interoperation with the ISDN networks in European countries like Italy, it is necessary to use this command to ensure normal interoperation, because the networks in this region have strict requirements on services and information elements.

Example

# Enable the Telecom Italia ISDN configuration.

[VG-Serial0:15] isdn communicate italy

4.1.10 isdn crlength

Syntax

isdn crlength { 1 | 2 }

View

cE1/PRI interface view

Parameter

1: Sets the call reference length to 1 byte.

2: Sets the call reference length to 2 bytes.

Description

Use the isdn crlength command to set the call reference length used when the cE1/PRI interface initiates calls.

Call reference is equivalent to the serial number allocated by the protocol to each call. In the length of 1 byte or 2 bytes, call reference is used cyclically.

If a VG interconnects with the peer device using the QSIG signaling, the VG always uniquely identifies a call by using the call reference field in the messages. According to the ECMA-143 protocol, the length of the call reference field can be either 1 byte or 2 bytes.

& Note:

When the VG receives a call from the peer end, the VG can automatically recognize the call reference length. However, some devices on the network cannot recognize the call reference length automatically. When the VG interconnects with such a device and initiates a call to it, the call reference length to be used by the VG must be configured to be consistent with that used by the peer device.

By default, the call reference length for the cE1/PRI interface is 2 bytes.

This command does not take effect if a call exists on the cE1/PRI interface, and must be carried out again when there is no call on the interface. Alternatively, you can disable the cE1/PRI interface manually by using the shutdown command, and enable it again by using the undo shutdown command after the configuration of this command. This operation will terminate the existing calls on the interface.

This command is valid only if the QSIG signaling is used on the cE1/PRI interface.

Related command: isdn protocol-type.

Example

# Set the call reference length used when calls are initiated from the cE1/PRI interface to 1 byte.

[VG-Serial0:15] isdn crlength 1

4.1.11 isdn facility-passthrough

Syntax

isdn facility-passthrough

undo isdn facility-passthrough

View

cE1/PRI interface view

Parameter

None

Description

Use the isdn facility-passthrough command to enable passthrough transmission of the facility information field of QSIG when the VG uses QSIG in interoperation with a PBX.

Use the undo facility-passthrough command to disable passthrough transmission of the facility information field of QSIG.

By default, passthrough transmission of the facility information field of QSIG is enabled.

Example

# Disable passthrough transmission of the facility information field of QSIG.

[VG-Serial0:15] undo isdn facility-passthrough

4.1.12 isdn ignore callednum

Syntax

isdn ignore callednum

undo isdn ignore callednum

View

cE1/PRI interface view

Parameter

None

Description

Use isdn ignore callednum command to disable the VG that is interoperating with an exchange to send SETUP ACK messages if the received SETUP messages in data service calls do not carry the called number information.

Use undo isdn ignore callednum command to enable the VG to send SETUP ACK messages in the same situation.

By default, the VG that is interoperating with an exchange sends SETUP ACK messages even if the received SETUP messages do not carry the called number information.

The switches of some vendors neither carry the called number information in the SETUP messages nor recognize SETUP ACK messages. In this case, a VG must be disabled to send SETUP ACK messages by using this command when interoperating with the switches of such vendors.

For a cCE1/PRI interface, you must perform PRI binding before you can configure this command in serial interface view. For more information about PRI binding, refer to the pri-set command (PRI Interface) in the interface part of this manual.

Example

# Disable the cE1/PRI interface on the VG from sending the SETUP ACK message.

[VG] controller e1 0

[VG-E1-0] pri-set

[VG-E1-0] interface serial 0:15

[VG-serial0:15] isdn ignore callednum

4.1.13 isdn ignore hlc

Syntax

isdn ignore hlc

undo isdn ignore hlc

View

cE1/PRI interface view

Parameter

None

Description

Use the isdn ignore hlc command to configure the SETUP message not to carry high-level compatible information unit when the ISDN originates calls. U

se the undo isdn ignore hlc command to restore the condition that the SETUP message carries high-level compatible information unit when the ISDN originates calls.

By default, the SETUP message carries high-level compatible information unit when the ISDN originates calls.

When interconnecting with European ISDN network, it is necessary to configure this command, that is, to configure the SETUP message not to carry high-level compatible information unit when the ISDN originates calls, because the European network cannot recognize the high-level compatible information unit.

Related command: isdn ignore llc.

Example

# When interconnecting with European ISDN network, if the European network cannot recognize the high-level compatible information unit, it is necessary to configure this command as follows.

[VG-Serial0:15] isdn ignore hlc

4.1.14 isdn ignore llc

Syntax

isdn ignore llc

undo isdn ignore llc

View

cE1/PRI interface view

Parameter

None

Description

Use the isdn ignore llc command to configure the SETUP message not to carry low-level compatible information unit when the ISDN originates calls.

Use the undo isdn ignore llc command to restore the condition that the SETUP message carries low-level compatible information unit when the ISDN originates calls.

By default, the SETUP message carries low-level compatible information unit when the ISDN originates calls.

When interconnecting with European ISDN network, it is necessary to configure this command, that is, to configure the SETUP message not to carry low-level compatible information unit when the ISDN originates calls, because the European network cannot recognize the low-level compatible information unit.

Related command: isdn ignore hlc.

Example

# When interconnecting with European ISDN network, if the European network cannot recognize the low-level compatible information unit, it is necessary to configure this command as follows.

[VG-Serial0:15] isdn ignore llc

4.1.15 isdn overlap-receiving

Syntax

isdn overlap-receiving

undo isdn overlap-receiving

View

cE1/PRI interface view

Parameter

None

Description

Use the isdn overlap-receiving command to set the mode in which the ISDN PRI interface receives called numbers to “overlap receiving”.

Use the undo isdn overlap-receiving command to set the mode in which the ISDN PRI interface receives called numbers to “complete receiving”.

By default, the overlap receiving mode is used.

If the calling side sends called numbers in the “overlap sending” mode, the number receiving mode of the interface should be set to “overlap receiving”; if the calling side sends called numbers in the “complete sending” mode, the number receiving mode of the interface can be set to “complete receiving”. This kind of corresponding relationship is maintained in accordance with this principle between the number receiving mode of the protocol interface and the number sending mode of the peer end.

However, when the calling side sends called numbers in the “complete sending” mode, the number receiving mode of the interface can also be set to the “overlap receiving” mode.

Related commands: isdn protocol-type, isdn sending-complete.

Example

# Set the mode in which the cE1/PRI PRI interface receives called numbers to “complete receiving”.

[VG-Serial0:15] undo isdn overlap-receiving

4.1.16 isdn protocol-type

Syntax

isdn protocol-type { dss1 | qsig }

View

cE1/PRI interface vie

Parameter

dss1: DSS1 (Digital Subscriber Signaling No.1) signaling is used.

qsig: QSIG signaling is used.

ni: NI(National ISDN) signaling is used.

Description

Use the isdn protocol-type command to configure signaling to be used at the cE1/PRI interface.

By default, DSS1 signaling is adopted.

When this command is used in Interface view, it can set the type of signaling for the active cE1/PRI interface. However, if there are calls on the interface, configuring this command will not take effect. In this case, you can modify the signaling type for the interface only after all the calls on the interface are disconnected or manually cleared by executing the commands shutdown and undo shutdown.

& Note:

l You cannot configure the command when there is active call on the cE1/PRI interface. Or you can use the shutdown command to disable the cE1/PRI interface to configure the command and then use the undo shundown command to enable the interface. This operation will disconnect the active calls on the interface.

l You are allowed to configure:

l The DSS1 protocol on cE1/PRI interfaces;

l QSIG on PRI interfaces.

l Other protocols are made up by the negotiation commands of Layer 3 protocol under DSS1 protocol.

Related command: display isdn call-info.

Example

# Configure the VG to use QSIG signaling on the cE1/PRI interface.

[VG-Serial0:15] isdn protocol-type qsig

4.1.17 isdn q931-timer

Syntax

isdn q931-timer timer-name time-interval

undo isdn q931-timer { timer-name | all }

View

cE1/PRI interface view

Parameter

timer-name: Name of Q931 timer. Refer to the following table for description in detail.

time-interval: Interval of timer. Refer to the following table for description in detail.

all: To be used to restore the default interval values of all the Q931 timers.

Description

Use the isdn q931-timer command to configure the interval for a Q931 signaling timer.

Use the undo isdn q931-timer command to restore the default interval values of Q931 signaling timers.

Different timers have different default values. Refer to the following table for description in detail.

When the DSS1 or QSIG protocol is enabled on an ISDN interface, you can customize the interval values of timers as needed. By executing the undo isdn q931-timer all command, you can restore the default interval values of all the Q931 timers.

Table 4-5 Description of Q931 timers

timer-name	Timer	Value range (in seconds)	Default value (in seconds)
t301	T301	30 to 1200	180
t302	T302	5 to 60	15
t303	T303	2 to 10	5
t304	T304	10 to 60	30
t305	T305	4 to 30	30
t308	T308	2 to 10	5
t309	T309	10 to 180	90
t310	T310	10 to 180	30
t313	T313	2 to 10	5
t316	T316	20 to 180	120
t322	T322	2 to 10	4

Normally, it is not necessary to change the values of Q931 signaling timers. If needed, please do that with the guide of qualified technical personnel.

Example

# Set the timer T322 to 6 seconds.

[VG-Serial0:15] isdn q931-timert 322 8

4.1.18 isdn qsig-timer

Syntax

isdn qsig-timer timer-name time-interval

undo isdn qsig-timer { timer-name | all }

View

cE1/PRI interface view

Parameter

timer-name: Name of QSIG timer. For details, see the table below.

time-interval: Interface of QSIG timer. For details, see the table below.

all: Restores the default interval setting of all QSIG timers.

Description

Use the isdn qsig-timer command to set the interval of the specified QSIG timer.

Use the undo isdn qsig-timer command to restore the default interval of the specified QSIG timer or all QSIG timers.

Different timers have different default values. For details, see the table below.

When the ISDN PRI interface uses the QSIG signaling, you can customize the timer settings based on your actual needs. The undo isdn qsig-timer all command can restore all the QSIG timers to their default settings.

Table 4-6 QSIG timer settings

timer-name	Timer	Value range (in seconds)	Default value (in seconds)
t301	T301	30 to 1200	180
t302	T302	5 to 60	15
t303	T303	2 to 10	5
t304	T304	10 to 60	20
t305	T305	4 to 30	15
t308	T308	2 to 10	5
t309	T309	10 to 180	90
t310	T310	10 to 180	30
t313	T313	2 to 10	5
t316	T316	2 to 180	5
t322	T322	2 to 10	5

Normally, it is not necessary to change the values of QSIG signaling timers. If needed, change the timer settings under the guide of qualified technical personnel.

Table 4-7 Description of QSIG signaling timers

Timer	Start condition	Stop condition
T301	Upon receiving ALERTING message	Upon receiving CONNECT message
T302	Starts upon sending SETUP ACK. Message, and restarts upon receiving INFORMATION message	Upon sending CALL PROCEEDING, ALERTING or CONNECT message
T303	Upon sending SETUP message	Upon receiving CALL PROCEEDING, CONNECT, ALERTING, SETUP ACK. or RELEASE COMPLETE message
T304	Upon receiving SETUP ACK. Message, or Upon sending INFORMATIONG message	Upon receiving CALL PROCEEDING, ALERTING or CONNECT message
T305	Upon sending DISCONNECT message	Upon receiving RELEASE or DISCONNECT message
T308	Upon sending RELEASE message	Upon receiving RELEASE or RELEASE COM. message
T309	When the data link layer is faulty and one or more calls are in the “ACTIVE: state	After reestablishment of the data link layer
T310	Upon receiving CALL PROCEEDING message	Upon receiving ALERTING, CONNECT, PROGRESS, DISCONNECT or RELEASE message
T313	Upon sending CONNECT message	Upon receiving CONNECT ACKNOWLEDGE message
T316	Upon sending RESTART message	Upon receiving RESTART ACKNOWLEDGE message
T322	Upon sending STATUS ENQUIRY message	Upon receiving STATUS, DISCONNECT, RELEASE or RELEASE COMPLETE message

This command is valid only when the cE1/PRI interface runs the QSIG signaling.

Related commands: isdn protocol-type, display isdn qsig-timer

Example

# Set the interval value of timer T322 to 6 seconds.

[VG-Serial0:15] isdn qsig-timer t322 6

# Restore the default values of all QSIG timers.

[VG-Serial0:15] undo isdn qsig-timer all

4.1.19 isdn sending-complete

Syntax

isdn sending-complete

undo isdn sending-complete

View

cE1/PRI Interface view

Parameter

None

Description

Use the isdn sending-complete command to enable a VG to carry the Sending-Complete Information Element in the SETUP message sent to PBX.

Use the undo isdn sending-complete command to disable the VG to carry the Sending-Complete Information Element in the SETUP message sent to PBX.

By default, the Sending-Complete Information Element is carried.

When a VG originates a call to PBX, information of the called number will be contained in the SETUP message and no INFORMATION message will be sent any more, regardless of whether the Sending-Complete Information Element is carried in the message.

This command cannot take effect unless the ISDN PRI interface adopts QSIG signaling.

Related command: isdn protocol-type, isdn overlap-receiving.

Example

# Disable the VG to carry the Sending-Complete Information Element in the SETUP message sent to PBX.

[VG-Serial0:15] undo isdn sending-complete

4.1.20 isdn service

Syntax

isdn service [ audio | speech ]

undo isdn service

View

cE1/PRI interface view

Parameter

audio: Sets the service type to “audio”.

speech: Sets the service type to “speech”.

Description

Use the isdn service command to set the service type supported by the cE1/PRI interface.

Use the undo isdn service command to remove all the service types supported by the cE1/PRI interface.

By default, the cE1/PRI interface supports all voice services.

If no parameter is specified, this command will set all the service types supported by the cE1/PRI interface.

Example

# Enable the cE1/PRI interface to support receiving audio type of services.

[VG-Serial0:15] isdn service audio

4.1.21 isdn waitconnectack

Syntax

isdn waitconnectack

undo isdn waitconnectack

View

cE1/PRI interface view

Parameter

None

Description

Use the isdn waitconnectack command to set VG so that it waits for the CONNECT ACK message, upon sending the CONNECT message, before it switches to the ACTIVE state and starts data and voice communication, when interoperating with a PBX.

Use the undo isdn waitconnectack command to set the VG so that it directly enter the ACTIVE state and starts data and voice communication, without waiting for the CONNECT ACK message.

By default, the VG waits for the CONNECT ACK message.

During the interoperation between VGs and PBXs, some PBXs respond with a CONNECT ACK message upon receiving the CONNECT message, and some do not. Therefore, for adaptation with different PBXs, it is necessary to specify whether the Q931 or QSIG protocol needs to wait for the CONNECT ACK message, upon sending the CONNECT message, before the VG switches to the ACTIVE state and starts data voice communication.

Example

# Configure the VG to enter the ACTIVE state without waiting for the CONNECT ACK message.

[VG-Serial0:15] undo isdn waitconnectack

Chapter 5 Voice RADIUS Configuration Commands

5.1 Voice RADIUS Configuration Commands

5.1.1 aaa-client

Syntax

aaa-client

View

Voice view

Parameter

None

Description

Use the aaa-client command to access the voice AAA service view.

Use the quit command to quit the view.

Related command: accounting, authentication-did, clienttype, and local-user.

Example

# Access the voice AAA service view.

[VG] voice-setup

[VG-voice] aaa-client

[VG-voice-aaa]

5.1.2 accounting

Syntax

accounting

undo accounting

View

Voice AAA service view

Parameter

None

Description

Use the accounting command to enable the voice call specific RADIUS accounting.

Use the undo accounting command to disable the function.

By default, voice call RADIUS accounting is disabled.

As for the same VG system providing voice call services, voice call RADIUS accounting is not only valid for two-stage dial but also valid for one-stage dial. Although authentication and authorization on two-stage and one-stage dial users are enabled separately, their accounting is enabled together. Enabling accounting on a VG system enables the accounting on all the calls regardless of whether they are from one-stage dial users or two-stage dial users. Likewise, disabling the accounting function on the VG disables the accounting on all calls.

& Note:

If you have enabled the accounting function by Use the accounting command, you must not configure the none keyword parameter in the acct-method command, thus ensuring the system can make accounting on VoIP calls.

Related command: acct-method, authentication-did, cdr, client-type, and local-user.

Example

# Enable the voice call RADIUS accounting function of the system.

[VG-voice-aaa] accounting

5.1.3 acct-method

Syntax

acct-method { none | start-ack | start-no-ack | stop-only }

undo acct-method

View

Voice AAA service view

Parameter

none: The VG system does not send accounting requests to the RADIUS Server, i.e., it does not implement accounting.

start-ack: At call setup, the VG starts a call connection only after receiving the accounting acknowledgement from the RADIUS Server, and it can release the call without receiving the response from the RADIUS server for the end-accounting request.

start-no-ack: The VG sends an accounting request message to the RADIUS Server when the call is set up and when the call ends. However, it directly initiates and releases the voice call no matter whether it has received a response from the RADIUS Server at the call setup and at the end of the call.

stop-only: The VG sends an end-accounting request only when the call ends and directly releases the call no matter whether it has received a response from the RADIUS Server.

Description

Use the acct-method command to apply an accounting method on the RADIUS Client (the VG system).

Use the undo acct-method command to restore the default accounting method.

By default, start-no-ack is adopted.

These accounting methods are introduced into the RADIUS Client (the VG system) intending for providing you with multiple choices. Thus, you can choose to control the access of IP calling subscribers by applying the start-ack method, by applying the stop-only method, or by applying the none method.

& Note:

If the keyword parameter none is specified, the system will make no accounting on any IP call subscribers, regardless of whether the accounting function (by using the accounting command) has been enabled.

Related command: accounting.

Example

# Adopt the start-ack accounting method.

[VG-voice-aaa] acct-method start-ack

5.1.4 authentication

Syntax

authentication

undo authentication

View

Access service ID view

Parameter

None

Description

Use the authentication command to enable user authentication specific to an access number.

Use the undo authentication command to disable the function.

By default, no authentication is enabled on users using any access number.

You can enable user authentication specific to an access number. If you do so, all the subscribers using the access number can obtain the authorization and place IP calls only if they pass the authentication. If you disable authentication, all the subscriber using the number will be able to place IP calls regardless of whether they are valid, i.e., the system will omit the authentication specific to the access number.

As authorization follows authentication, disabling the authentication specific to an access number also disables the authorization specific to the number.

Related command: gw-access-number, authorization, accounting, and authentication-did.

Example

# Make authentication on the subscribers using the two-stage dial access number 18901.

[VG-voice-dial-anum18901] authentication

5.1.5 authentication-did

Syntax

authentication-did

undo authentication-did

View

Voice AAA service view

Parameter

None

Description

Use the authentication-did command to enable authentication on all the one-stage dial subscribers.

Use the undo authentication-did command to disable the function.

By default, authentication is not enabled on one-stage dial subscribers.

This command is valid for one-stage subscribers (those who can directly dial destination numbers) but not for two-stage dial subscribers (those who must dial the access numbers before dial destination numbers).

Executing this command enables the authentication on all the one-stage dial subscribers. The calling number of each one-stage IP call subscriber will be extracted and sent to the RADIUS Server for authentication. If the subscriber passes the authentication, he will get the subsequent service. If not, the connection will be disconnected and the subscriber will be unable to place VoIP calls.

Disabling this function also disables the authorization on one-stage dial subscribers.

Related command: authorization-did and accounting.

Example

# Enable the authentication on one-stage dial subscribers.

[VG-voice-aaa] authentication-did

5.1.6 authorization

Syntax

authorization

undo authorization

View

Access service ID view

Parameter

None

Description

Use the authorization command to enable user authorization specific to an access number.

Use the undo authorization command to disable the function.

By default, no authorization is enabled on users using any access number.

You can enable user authorization specific to an access number.

Only when authentication on two-stage dial subscribers is enabled, can this command take effect and can the authorization specific to an access number be enabled. In other words, authorization is implemented on the premise of the enabling of authentication.

& Note:

Authorization is a concept different from authentication. Authentication, authorization, and accounting are commonly referred to as access-user specific AAA.

l As for a subscriber, authentication is the verification on his identity by asking the question “who's this?”. The essential concern underlying the question is whether the subscriber has been granted any right.

l Authorization makes sure the rights that the subscriber has been granted by asking the question “What are the rights?”. The essential concern underlying the question is the range of the rights that the subscriber enjoys.

l In the voice RADIUS application, voice RADIUS determines to allow/disallow a subscriber to place VoIP calls based on the provided user information such as calling number and card Number/PIN at the AUTHETNICATION stage; it determines the type of VoIP call that the subscriber can place based on the right range and the request (called number) at the AUTHORIZATION stage, whether it is a local call, a national toll call, or an international toll call.

Related command: gw-access-number, authentication, accounting, and authorization-did.

Example

# Authorize the subscribers using the access number 18901.

[VG-voice-dial-anum18901] authentication

[VG-voice-dial-anum18901] authorization

5.1.7 authorization-did

Syntax

authorization-did

undo authorization-did

View

Voice AAA service view

Parameter

None

Description

Use the authorization-did command to enable authorization on all the one-stage dial subscribers.

Use the undo authorization-did command to disable the function.

By default, authorization on one-stage dial subscribers is not enabled.

This command is valid for one-stage subscribers but not for two-stage dial subscribers.

This command will not take effect unless the authentication on one-stage dial subscribers has been enabled.

Executing this command enables the authorization function. All the one-stage subscribers attempting to place IP calls will be authenticated and if they pass the authentication, their dialed numbers will be sent to the RADIUS Server for authorization. If a subscriber passes the authorization, his dialed number will be used for searching for the IP address of the called party and he will be able to get through the IP call. Otherwise, the connection for the subscriber will be disconnected and no IP call can be put through.

Related command: authentication-did and accounting.

Example

# Enable authorization on one-stage dial subscribers.

[VG-voice-aaa] authorization-did

5.1.8 callednumber

Syntax

callednumber receive-method { immediate | terminator }

undo callednumber receive-method

View

Access service number view

Parameter

receive-method: Mode to receive the called number. By default, the mode to receive the called number is terminator.

immediate: Initiate a call as soon as collecting all the numbers.

terminator: Terminates the called number with #.

Description

Use the callednumber command to configure the mode to receive the called number.

Use the undo callednumber command to restore the default mode.

In the two-stage dialing process, by default, you need to enter the # to terminate the dialing of the called number. Use the command to configure to initiate a call after collecting all the number without entering the #.

Example

# Configure the mode to receive the called number to immediate call for the access service number 17990.

[VG-voice-dial-anum17990] callednumber receive-method immediate

5.1.9 card-digit

Syntax

card-digit card-digit

View

Access service ID view

Parameter

card-digit: Number of digits of a card number, which is in the range of 1 to 31 and defaults to 12.

Description

Use the card-digit command to configure the number of digits of a card number for an access number.

This command is used for configuring the number of digits of a card number applied in a two-stage dial card process in which card number/PIN is used for authentication. Once the number of digits of a card number has been specified for an access number, any subscriber using that number must input a card number compliant with the number of digits provided for.

This command can take effect only if you have specified the dial process for the current access number to be card process (by suing the process-config command). Otherwise, it will be rendered useless.

Related command: gw-access-number, process-config, and password-digit.

Example

# Set the number of digits of card numbers for the access number 18901 to 10.

[VG-voice-dial-anum18901] card-digit 10

5.1.10 cdr

Syntax

cdr { buffer [ size-number ] | duration [ timer-number ] | threshold [ percentage ] }

undo cdr

View

Voice AAA service view

Parameter

buffer [ size-number ]: Limits the number of retained Call Detail Records (CDRs) by putting an upper limit on the size of retained CDRs. The value of size-number is in the range of 0 to 500 and defaults to 50, with 0 indicating that no CDR is retained.

duration [ timer-number ]: Limits the number of retained CDRs by specifying the lifetime of CDRs starting from the termination of sessions. The timer-number argument is the lifetime of CDRs in seconds. It is in the range of 0 to 2147483647 and defaults to 86400 (namely, 24 hours), with 0 indicating that no CDR is retained.

threshold [ percentage ] : Provides warning about the call CDRs generated based on the specified output threshold (in percentage) of CDRs. Presently, the VG supports warning through the output mode of the information center. In this parameter, percentage is the threshold value. The value range is 0 to 100 and the default value is 80% of the maximum number of retained CDRs. For example, 500 ´ 80% = 400 CDRs.

Description

Use the cdr command to configure the rule observed by the system for CDR retention.

Use the undo cdr command to restore the default CDR retention rule.

The system will retain CDRs according to the rule you have configured using this command.

& Note:

Comware allows the retention of up to 500 CDR entries.

To put it specific, even if you have specified the allowed longest CDR lifetime for retention, the system cannot retain CDR entries more than 500. In case more than 500 CDR entries are created in compliance with the configured lifetime as a result of the generation of enormous traffic during a period of time, those terminated earlier and beyond 500 will be deleted, despite they are compliant with the retention rule.

Related command: display aaa unsent-h323-call-record and display voice call-history-record.

Example

# Allow the system to retain up to 400 CDR entries.

[VG-voice-aaa] cdr buffer 400

# Allow the system to retain the CDRs of the calls terminated within the last ten minutes.

[VG-voice-aaa] cdr duration 10

# Set the threshold of CDRs to 10.

[VG-voice-aaa] cdr threshold 10

5.1.11 clienttype

Syntax

clienttype { overload-nonstandard | vsa-nonstandard | private | ietf-rfc }

undo clienttype

View

Voice AAA service view

Parameter

overload-nonstandard: Intends for the interoperation with non-standard RADIUS servers in the industry. In this mode, the extended accounting information is encapsulated in the RFC provisioned attribute “Acct-Session-Id”.

vsa-nonstandard: Intends for the interoperation with non-standard RADIUS servers in the industry. In this mode, the extended accounting information is encapsulated in the RFC provisioned attribute “Vendor-Specific Attributes”, allowing for a wider range of private properties. To use this parameter, the server-authorization command must be executed.

private: Applies the private protocol on the RADIUS Client. For example, use the device together with a H3C CAMS server.

ietf-rfc: Intends for the interoperation with standard RADIUS Servers, i.e., the servers in strict compliance with RFC2865 and RFC2866.

Description

Use the clienttype command to apply the RADIUS protocol type (in extended format) on the RADIUS Client for authentication, authorization, and accounting.

Use the undo clienttype command to restore the default protocol type.

By default, the RADIUS Client adopts the H3C (private) protocol.

Different voice gateway (RADIUS Client) vendors and RADIUS Server providers have different implementation schemes in voice RADIUS AAA implementation. Naturally, the use and allocation of the private attribute fields in the RADIUS protocols are different.

So far, Comware supports four RADIUS Client protocol types, allowing for the interoperation with the RADIUS Servers of most vendors.

Related command: server-authorization

Example

# Apply vsa-nonstandard on the RADIUS Client.

[VG-voice-aaa] clienttype vsa-nonstandard

5.1.12 debugging voice data-flow

Syntax

debugging voice data-flow { all | detail | error | fax [ error ] | jitter [ error ] | receive | send | vpp }

View

Any view

Parameter

all: Enables all debugging of voice fax data-flow.

detail: Enables debugging of the detailed voice data-flow.

error: Enables error debugging of voice data-flow.

fax: Enables debugging of fax data-flow.

jitter: Enables debugging of jitter-buffer processing.

receive: Enables debugging of receiving voice data-flow.

send: Enables debugging of sending voice data-flow.

vpp: Enables debugging of voice data-flow vpp processing.

Description

Use the debugging voice data-flow command to enable the debugging of the voice fax data-flow module.

Example

# Enable the debugging of voice fax data-flow.

[VG] debugging voice data-flow fax

5.1.13 debugging voice radius

Syntax

debugging voice radius { event | packet | primitive }

View

Any view

Parameter

event: Enables debugging of RADIUS events.

packet: Enables debugging of receiving and sending RADIUS packets.

primitive: Enable debugging of RADIUS primitive.

Description

Use the debugging voice radius command to enable RADIUS debugging. Use the undo debugging voice radius command to disable RADIUS debugging.

Example

# Enables debugging of RADIUS events.

[VG] debugging voice radius event

5.1.14 debugging voice vcc

Syntax

debugging voice vcc { all | error | ipp | proc | radius | rcv | timer | vpp | channel channel-number }

View

Any view

Parameter

all: Enables all debugging of the VCC software module.

error: Enables error debugging of the VCC software module.

ipp: Enables debugging on the messages between the VCC software module and the IPP module.

proc: Enables debugging on the messages between the VCC software module and the system process.

radius: Enables debugging on the messages between the VCC software module and the RADIUS Client.

rcv: Enables debugging on the messages between the VCC software module between the RCV module.

timer: Enables debugging on the messages between the VCC software module between the timer module.

vpp: Enables debugging on the messages between the VCC software module between the VPP module.

channel [ channel-number ]: Enables debugging on the specified channel of the VG. The parameter channel-number specifies a channel and its value range depends on the type and number of voice cards actually in use.

Description

Use the debugging voice vcc command to enable, and specify different levels and types of, debugging on the VCC software module.

The debugging voice vcc channel channel-number command and the debugging voice vcc all command mutually independent. By using the debugging voice vcc channel channel-number command, you can enable only debugging for the specified channel, without enabling the specific debugging information output. To view the specific debugging information, you need to use this command with other debugging information switches; the debugging voice vcc all can enable all the debugging switches of VAS except channel debugging.

The undo debugging voice vcc all command and the undo debugging voice vcc channel command are completely independent of each other. The execution of the undo debugging voice vcc all command will disable all the debugging switches including channel debugging, which is the result of the undo debugging voice vcc channel command, while the undo debugging voice vcc channel command disables only channel debugging, without disabling other debugging switches of the VAS module.

& Note:

When specifying debugging level/type, you should note that:

l The parameter channel-number could not be replaced by any character rather than a number. A channel-number containing characters other than a number is invalid.

l The specified channel number must not be greater than the number of available channels. Otherwise, the input number will be invalid.

Related command: reset voice vcc and display voice vcc.

Example

# Enables debugging on the messages sent from the VCC software module to the RADIUS Client.

[VG] debugging voice vcc radius

5.1.15 display aaa unsent-h323-call-record

Syntax

display aaa unsent-h323-call-record

View

Any view

Parameter

None

Description

Use the display aaa unsent-h323-call-record command to view the CDRs of unsent calls.

In the event that you apply the start-ack accounting mode on the RADIUS Client (VG system), the system will be unable to set up the VoIP-side session for a call if the RADIUS Server does not return the acknowledgement for the call within the specified time period. The system will retain the information of the unsent call and will initiate another accounting request to the RADIUS Server for a connection. This command is used for displaying CDRs of the unsent calls due to the VoIP-side session setup failure as a result of the failure in receiving the START-ACK messages from the RADIUS Server. You can make use of the output information of this command to isolate faults and make troubleshooting.

Example

# Display the CDRs of the unsent calls due to the failure in setting up VoIP-side sessions as a result of the failure in receiving START-ACK messages from the RADIUS Server.

[VG] display aaa unsent-h323-call-record

Index = 1

Acct_session_Id = 10

CallOrigin = Answer

CallType = Telephony

Callernumber = 1000

Callednumber = 1001

CallDuration = 00:00:03

TransmitPackets = 1000

TransmitBytes = 32000

ReceivePackets = 1100

ReceiveBytes = 35200

Table 5-1 Description on the fields of the display aaa unsent-h323-call-record command

Field	Description
Index	Index of the unsent call record
Acct_session_Id	RADIUS session ID determined by the negotiation between the RADIUS Server and the RADIUS Client
CallOrigin	Call direction, identifying the local end as the calling party or the called party
CallType	Call type that can be either telephony or fax
Callernumber	Calling number
Callednumber	Called number
CallDuration	Call duration
TransmitPackets	Packets transmitted from the local end
TransmitBytes	Bytes transmitted from the local end
ReceivePackets	Packets received at the local end
ReceiveBytes	Bytes received at the local end

5.1.16 display current-configuration voice

Syntax

display current-configuration voice [ aaa | access-number | cdr | acct-method ]

View

Any view

Parameter

aaa: Displays the AAA configuration information.

access-number: Displays the configuration information related to access numbers.

cdr: Displays the CDR retention rule information.

acct-method: Displays the accounting method configuration.

Description

Use the display current-configuration voice command to display the current voice RADIUS configuration in the system.

By default, executing the command without the parameters aaa, access-number, cdr, and acct-method displays all the configuration information.

Example

# Display the current voice RADIUS configuration.

[VG] display current-configuration voice aaa

AAA ( accounting authentication authorization ):

accounting = off

authentication-did = off

authorization-did = off

[VG] display current-configuration voice acct-method

accounting-method = start-no-ack

[VG] display current-configuration voice access-number

accessnumber (caller):

#access number: [ 17900 ]

authentication = off

authorization = off

accessnumber ( card ):

#access number: [ 17907 ]

authentication = off

authorization = off

card digit = 12

password digit = 6

redial times = 3

[VG] display current-configuration voice cdr

dial-control-mib-info:

buffer = 50

duration = 15

Table 5-2 Description on the fields of the display current-configuration voice command

Field	Description
accounting	Enables/Disables accounting.
authentication-did	Enables/Disables one-stage dial user authentication.
authorization-did	Enables/Disables one-stage dial user authorization.
accounting-method	Accounting method
accessnumber	Access number type divided basing on dial process
access number	Access number
authentication	Enables/Disables authentication specific to an access number.
authorization	Enables/Disables authorization specific to an access number.
card digit	Card digits of access number
password digit	PIN digits of access number
Redial times	Redial times that an access number allows
buffer	The maximum size of the retained CDR
duration	The maximum CDR retention duration

The above information indicates that two access service numbers have been configured, including: The calling number process (17900), no authentication, no authorization; the card number process (17907), no authentication, no authorization, the card number digits are 12, the password digits are 6, and each number can be dialed for 4 times (i.e., the card number, password and called number all can be redialed for 3 times). In the end, the information displays the call record capacity of the dial-control-MIB and the record retaining time.

5.1.17 display voice aaa-client local-user

Syntax

display voice aaa-client local-user

View

Any view

Parameter

None

Description

Use the display voice aaa-client local-user command to display the user names and passwords in the local voice user database (for voice users only).

Example

# Display information in the local voice user database.

[VG] display voice aaa-client local-user

Current voip user number : 1

Username : 1234567890 Password : 445566 Status : valid

Table 5-3 Description on the fields of the display voice aaa-client local-user command

Field	Description
Current voip user number	Number of VoIP users
Username	Voice user name
password	VoIP user password
status	VoIP user status

The information above means that the local voice user database contains one user, whose use name is 1234567890 and password is 445566.

5.1.18 display voice aaa-client statistic

Syntax

display voice aaa-client statistic

View

Any view

Parameter

None

Description

Use the display voice aaa-client statistic command to view the statistics related to the RADIUS Server.

Executing this command can display the offline duration of the RADIUS Server and the CDR entries thus lost. (Offline duration refers to the duration in which the RADIUS Server fails in providing services or has no response.)

Related command: reset voice aaa-client statistic.

Example

# Display the statistics related to the RADIUS Server.

[VG] display voice aaa-client statistic

Server down duration : 00h 33m 30s

The VoIP dropped end-accounting packets : 356

Table 5-4 Description on the fields of the display voice aaa-client statistic command

Field	Description
Server down duration	RADIUS server down time
The VoIP dropped end-accounting packets	Dropped en-accounting request packets

The information above means that the down time of communication with the RADIUS server is 33 minutes and 30 seconds, and the 356 end-accounting request packets are dropped.

5.1.19 display voice call-history-record

Syntax

display voice call-history-record { callednumber called-number | callernumber caller-number | cardnumber card-number | remote-ip-addr ip-address | last last-number } [ brief ]

View

Any view

Parameter

callednumber called-number: Displays CDR information by called number, with called-number being an E.164 number comprising less than 31 characters.

callernumber caller-number: Displays CDR information by caller number, with caller-number being an E.164 number comprising less than 31 characters.

cardnumber card-number: Displays CDR information by prepaid card number, with card-number being a string comprising less than 31 characters.

remote-ip-addr ip-address: Displays CDR information by the IP address of called party, with ip-address being the IP address of called party in dotted decimal format.

last last-number: Displays CDRs of the specified number of last calls, with last-number in the range of 1 to 500.

brief: Displays CDR information in brief format.

Description

Use the display voice call-history-record command to view the CDR information.

& Note:

When configuring the filtering rule for display, you should note that:

l The specified called-number/caller-number must not contain unacceptable characters. Your input effort will be invalid if you input any character other than a digit, “0 to 9”, “*”, “#”, “T”, or “.”;

l The last-number and line number arguments must not contain characters other than numerals; otherwise, your input effort will be invalid;

l The specified called-number/caller-number must not exceed 31 characters. A called-number or caller-number comprising more than 31 characters is invalid;

l The value of last-number must not exceed 500; otherwise, your input effort will be invalid;

l When a call setup fails, the call information will also be recorded, and the call attributes that cannot be identified before the call failure will be replaced with the default values.

The reason of the RADIUS Server sends the acknowledgement:

l Normal release

l Card number not exist

l Invalid password

l This accounts is using

l No enough balance

l The accounts is expired

l Credit limit

l User reject

l Service invalid

l Called limit

Related command: display voice call-history-record line.

Example

# Filter by calling number to display the history record of the last ten calls.

[VG] display voice call-history-record last 10

CallRecord[30]:

CallerNum = 4000

CalledNum = 2000

EncodeType = 711u

PeerAddress = 127.0.0.1

DisconnectCause = 0

DisconnectText = Normal release

TalkingTimes = 00h 00m 00s

VoiceTimes = 00h 00m 00s

FaxTimes = 00h 00m 00s

ImgPages = 0

CallDirection = 2

SetupTime(voip) = Mar 11, 2000 02:20:27

ConnectTime(voip) = None

DisconectTime(voip) = Mar 11, 2000 02:20:29

Transmit (voip) = 0 (package) : 45456 (byte)

Received(voip) = 0 (package) : 45456 (byte)

SetupTime(pstn) = None

ConnectTime(pstn) = None

DisconectTime(pstn) = None

Transmit (pstn) = 0 (package) : 45456 (byte)

Received(pstn) = 0 (package) : 45456 (byte)

Table 5-5 Description on the fields of the display voice call-history-record command

Item	Description
CallRecord[30]	Number of call record
CallerNum	Calling number
CalledNum	Called number
EncodeType	Encoding type
PeerAddress	Peer address
DisconnectCause	Code of disconnect cause
DisconnectText	Text of disconnect cause
TalkingTimes	Talking duration
VoiceTimes	Voice talking duration
FaxTimes	Fax duration
ImgPages	Facsimile pages
CallDirection	Calling direction
SetupTime(voip)	Call setup time on network side
ConnectTime(voip)	Call connecting time on network side
DisconectTime(voip)	Call disconnecting time on network side
Transmit (voip)	Number of packets transported on network side (packet/byte)
Received(voip)	Number of packets received on network side (packet/byte)
SetupTime(pstn)	Call setup time on PSTN side
ConnectTime(pstn)	Call connecting time on PSTN side
DisconectTime(pstn)	Call disconnecting time on PSTN side
Transmit (pstn)	Number of packets transported on PSTN side (packet/byte)
Received(pstn)	Number of packets received on PSTN side (packet/byte)

5.1.20 display voice vcc

Syntax

display voice vcc { channel [ channel-number ] | statistic { all | error | ipp | proc | rcv | rds | timeout | vpp } }

View

Any view

Parameter

channel [ channel-number ]: Displays the call channel state information of the channel specified by the parameter channel-number. The value range for this parameter depends on the type and number of voice cards actually in use.

statistic: Displays the statistics of the VCC software module.

all: Displays all the statistics of the VCC software module.

error: Displays the error statistics of the VCC software module.

ipp: Displays the statistics of the messages for the interaction between the VCC and the IPP software modules.

proc: Displays the statistics of the messages for the interaction between the VCC software module and the system process.

rcv: Displays the statistics of the messages for the interaction between the VCC and the RCV software modules.

rds: Displays the statistics of the messages for the interaction between the VCC and the RDS software modules.

timeout: Displays the statistics of the messages for the interaction between the VCC software module and the timer module.

vpp: Displays the statistics of the messages for the interaction between the VCC and the VPP software modules.

Description

Use the display voice vcc command to view the call channel state information and the call statistics.

By default, executing this command without the parameter channel-number displays the state information of all channels.

& Note:

When configuring the filtering rule for display, note that the specified channel number must not be greater than the number of available channels. Otherwise, your input effort will be rendered useless.

Related command: reset voice vcc and debugging voice vcc.

Example

# Displays the call state information on channel 4 of the VG.

[VG] display voice vcc channel 4

CallRecord [20394384]:

Index = 34

CallerNum = 4000

CalledNum = 4940

EncodeType = 729

PeerAddress = 192.168.80.30

TalkingTimes = 00h 00m 01s

VoiceTimes = 00h 00m 01s

FaxTimes = 00h 00m 00s

ImgPages = 0

CallDirection = 2

VoIP:

SetupTime(voip) = Jan 1, 1998 09:04:44

ConnectTime(voip) = Jan 1, 1998 09:04:44

Transmit(voip) = 12 (package): 20394224 (byte)

Received(voip) = 12 (package): 20394224 (byte)

PSTN:

SetupTime(pstn) = Jan 1, 1998 09:04:44

ConnectTime(pstn) = Jan 1, 1998 09:04:44

Transmit(pstn) = 12 (package): 20394224 (byte)

Received(pstn) = 12 (package): 20394224 (byte)

Table 5-6 Description of call channel status

Item	Description
Index	Index of current call in channel
CallerNum	Caller number of current call in channel
CalledNum	Called number of current call in channel
EncodeType	Compression/Decompression type of current call in channel
PeerAddress	IP address of Peer GW of current call in channel
TalkingTimes	Call duration of current call in channel
VoiceTimes	Total voice duration of current call in channel at PSTN side
FaxTimes	Total fax duration of current call in channel at PSTN side
ImgPages	Total pages of sending/receiving fax
CallDirection	Direction of current call in channel, it means local end is caller end or called end corresponding with 1 or 2 respectively
SetupTime(voip/pstn)	Time of current call is setup(VoIP/PSTN)
ConnectTime(voip/pstn)	Connect start time of current call in channel(VoIP/PSTN)
Transmit	Number of packets and number of bytes of the packets sent at local end of current call in channel
Receive	Number of packets and number of bytes received at local end of current call in channel

5.1.21 gw-access-number

Syntax

gw-access-number access-number

undo gw-access-number [ access-number ]

View

Voice dial-program view

Parameter

access-number: The specified access number (such as 169 and 17900), a string comprising a maximum of 31 characters, which can be numerals and wildcard “.”. The character string range is ^([0-9]+.%)$. The wildcard “.” can be replaced by any character. The wildcard can only be put at the end of the string. You can configure a maximum of 100 access numbers.

Description

Use the gw-access-number command to configure an access number or access the access number view.

Use the undo gw-access-number command to delete one or all the access numbers.

You can delete a single access number by specifying the parameter access-number in the undo form of the command. If you do not specify the parameter, the system will delete all the configured access numbers.

Before deleting all the access numbers, the system displays the following operation warning:

Delete all of the access number? (n/y)

You can press <Y> to confirm the deletion or <N> to cancel the deletion operation.

& Note:

When configuring an access number, you should note that:

l The wildcard “.” Must not appear at the beginning or in the middle of the specified access number, but it can appear at the end. For example, “.1234”, “123..4” and “…” are all invalid access numbers, while “1234..” is a valid access number.;

l A number that completely or partially match with an existing access number is also an invalid access number. For example, if “1234” already exists, you cannot specify “123” or “123456” as an access number.

l The specified access-number cannot exceed 31 characters. An access-number comprising more than 31 characters is invalid.

l The system allows up to 100 access numbers. You will be unable to add new numbers if there have been 100 access numbers and must delete some numbers before you can add new ones.

l The voice entity and access number configuration policy of the system is as follows: if number-match longest is used, the system will perform access number match first; if number-match shortest is used, the shortest number will be matched first, and if they have the same length, the system will perform access number match first.

Related commands: process-config, card-digit, password-digit, redialtimes.

Example

# Configure the access number 18901 and access the access number view.

[VG-voice-dial] gw-access-number 18901

[VG-voice-dial-anum18901]

# Delete all the configured access numbers from the system.

[VG-voice-dial] undo gw-access-number

5.1.22 local-user (aaa-client)

Syntax

local-user username [ password password ]

undo local-user username

View

Voice AAA service view

Parameter

username: Username in the range of 1 to 16 characters. Allowed characters are 0-9#*.

password password: User authentication password in the range of 1 to 16 characters. Allowed characters are 0-9#*. The user password will be left blank if no value is input.

Description

Use the local-user command to add a username and its password into the local voice user database.

Use the undo local-user command to delete the specified local voice user.

By default, no local voice user and password is configured.

If you have configured a local voice user database, the system perform local-first authentication on voice subscribers. If passing the local authentication, a voice subscriber will not have to undergo the subsequent RADIUS authentication. If not, he will undergo the authentication conducted by the RADIUS Server. Only if the RADIUS authentication succeeds will the subscriber be allowed to carry out voice communications. So far, the local voice user database is allowed to accommodate 200 local users.

Medium- and large-sized enterprises are recommended to directly adopt RADIUS authentication rather than configuring local-first authentication

Related command: clienttype.

Example

# Configure a local voice user, given the username is “8801456302” and the password is “30017”.

[VG-voice-aaa] local-user 8801456302 password 30017

5.1.23 password-digit

Syntax

password-digit password-digit

View

Access service ID view

Parameter

password-digit: Number of password digits, which is in the range of 1 to 16 and defaults to 6.

Description

Use the password-digit command to configure the number of password digits for the card process of an access number.

This command is used to configure the number of password digits for the two-stage dial card process in which card number/PIN is used for authentication.

You must specify the dial process of the access number to card process by using the process-config command before you configure this command. Otherwise, the configuration of this command will be rendered useless and the access number cannot be retrieved at all.

Related command: gw-access-number, process-config, and card-digit.

Example

# Set the number of password digits to 4 for the access number “18901”.

[VG-voice-dial-anum18901] password-digit 4

5.1.24 process-config

Syntax

process-config { callernumber | cardnumber | voice-caller }

View

Access service ID view

Parameter

callernumber: Calling number dial process. In this mode, the system sends the dial tone to subscribers after they dial access numbers so that they can input the destination numbers. In this flow, user authentication is accomplished by identifying calling number.

cardnumber: Card number dial process. In this mode, the system plays messages after subscribers dial access numbers, prompting them to input prepaid card numbers and PINs. In this process, user authentication is accomplished by identifying prepaid card number/PIN or local user card number/PIN. Accounting cannot be performed in local authentication mode. The VG plays a voice announcement to users about the balance in their accounts, and the maximum balance amount is 999999 dollars and 99 cents.

voice-caller: Voice caller number process. Configured with it, the system plays voice messages to users after they dial the access number, asking them to select language and input the called numbers. In this process, users are authenticated using calling number identification (CNI).

When the user changes the current dial process (that is, between the calling number dial process and the card number dial process), the system will restore the involved parameters (for example, card number/PIN, and redial times in the card number process) to the defaults automatically.

Description

Use the process-config command to specify a dial process for an access number.

By default, card number process is applied to all the access numbers.

Each access number has a dial process for it and all its subscribers must set up calls following the same process.

In Comware, three dial processes are available: calling number process, card number process, and voice caller number process.

l Calling number process: It is actually the calling number authentication process. It is thus named because the system implements authentication, authorization, and accounting based on the calling numbers of subscribers. In this process, you do not need to make further configuration on the parameters in the process.

l Card number process: As the name implies, a subscriber using the card number process needs to dial his card number and password after dialing the access number in order to finish the authentication process. Only after the subscriber passes the authentication can he dial the destination number. In the card number process, you can use the commands card-digit, password-digit, and redialtimes to configure the parameters of the process.

l Voice caller number process: uses calling numbers for authentication. After a user dials the access number, the voice gateway uses the calling number to authenticate the user. If authentication is passed, the voice gateway plays a voice message to prompt the user to select a prompt message language and dial the called number. This process allows users to enable/disable prompt language selection and configure the allowed number of the second-stage dial attempts.

This is how the voice caller number process differs from the calling number dial process:

In the voice caller number process the voice gateway plays voice messages to prompt you to select a language and input the called number after you dial the access number; but in the calling number dial process it only plays dial tone (long tone).

Related commands: gw-access-number, card-digit, password-digit, redialtimes.

Example

# Specify the subscriber access process for the access number 18901 to card number process.

[VG-voice-dial-anum18901] process-config cardnumber

5.1.25 radius retry

Syntax

radius retry times

undo radius retry

View

System view

Parameter

times: Times of retransmitting request packet to RADIUS, ranging 1 to 255. The default value is 3 times.

Description

Use the radius retry command to configure the times of retransmitting request packet to RADIUS server.

Use the undo radius retry command to restore the default value.

If no acknowledgement is received from the RADIUS server within the timeout value after an AAA request is sent to the RADIUS server, it is necessary to retransmit the AAA request. If the number of AAA request times exceeds the specified number of times, it is deemed that this server can not work normally any more.

Example

# Configure the number of retransmitting request packet to RADIUS server to 2 times.

[VG] radius retry 2

5.1.26 radius server

Syntax

radius server { hostname | ip-address } [ authentication-port port-number ] [ accounting-port port-number ]

undo radius server { hostname | ip-address }

View

System view

Parameter

hostname: The RADIUS server name.

ip-address: The RADIUS server ip address (in dotted decimal)

authentication-port: Authentication port number specified. The default value is 1812.

accounting-port: Accounting port number specified. The default value is 1813.

port-number: Monitoring port number of the RADIUS server. 0 indicates that the client will not use the authentication or accounting function provided by the server.

Description

Use the radius server command to configure the host IP address (or hostname) of the RADIUS server, authentication port number and accounting port number.

Use the undo radius server command to cancel the RADIUS server with a specified host IP address or hostname.

Multiple RADIUS servers can be configured by execute this command many times (3 servers can be configured at most). System select RADIUS server by following the principles:

l First configured first selected.

l When the first server does not respond, the next server is selected in sequence.

Example

# Configure the host with the IP address 129.102.0.2 as an authentication server only, and the authentication port number as 1000.

[VG] radius server 129.102.0.2 authentication-port 1000 accounting-port 0

5.1.27 radius shared-key

Syntax

radius shared-key string

undo radius shared-key

View

System view

Parameter

string: Key of the RADIUS server, ranging 1 to 16 characters.

Description

Use the radius shared-key command to configure the key for the RADIUS server.

Use the undo radius shared-key command to cancel the key of the RADIUS server.

The key is used to encrypt the user’s password and generate the Response Authenticator. It is required that the configured key should be consistent with the one of RADIUS server.

Example

# Configure the key of the RADIUS server as abc.

[VG] radius shared-key abc

5.1.28 radius source-ip

Syntax

radius source-ip ip-address

undo source-ip

View

System view

Parameter

ip-address: The source IP address of all the RADIUS packets.

Description

Use the radius source-ip command to specify a source IP address for the RADIUS packets transmitted from the VG.

Use the undo radius source-ip command to disable specifying a source IP address for the RADIUS packets.

By default, no IP address is specified for transmitting RADIUS packets

If the same source IP address is specified for all the RADIUS packets, the RADIUS server will only need to contact the VG using that IP address, instead of registering the IP addresses of all the interfaces that are likely to send RADIUS packets.

Example

# Use 192.168.80.1 as the source IP address of all the RADIUS packets.

[VG] radius source-ip 192.168.80.1

5.1.29 radius stop-resend

Syntax

radius stop-resend times

undo radius stop-resend

View

System view

Parameter

times: The maximum number of times that the VG is allowed to transmit the RADIUS voice accounting STOP packets, which is in the range of 30 to 2000 and defaults to 100.

Description

Use the radius stop-resend command to configure the maximum number of times that the VG is allowed to transmit the RADIUS voice accounting STOP packets.

Use the undo radius stop-resend command to restore the default maximum number of times that the VG is allowed to transmit the RADIUS voice accounting STOP packets.

This command is only useful for the RADIUS accounting of voice and will not affect any other types of accounting.

Example

# Set 200 as the maximum number of times that the VG is allowed to transmit the RADIUS voice accounting STOP packets.

[VG] radius stop-resend 200

5.1.30 radius timer quiet

Syntax

radius timer quiet minutes

undo radius timer quiet

View

System view

Parameter

minutes: Quiet period after the server fails, in minutes, ranging 1 to 255. The default value is 5 minutes.

Description

Use the radius timer quiet command to configure the invalid time period after the RADIUS server fails.

Use the undo radius timer quiet command to restore the default value of the invalid time period.

When the RADIUS server fails (like the line between NAS and RADIUS has a loosened screw or the RADIUS process fails to function), the system will set its status as down. After the resuming time configured above, the system will set its status as up. If the server in service now fails, the system will auto-check whether or not the original server can be put into service.

Example

# Configure 10 minutes as the invalid time period after the failure of the RADIUS server.

[VG] radius timer quiet 10

5.1.31 radius timer realtime-accounting

Syntax

radius timer realtime-accounting minutes

undo radius timer realtime-accounting

View

System view

Parameter

minutes: Timeout value for real-time accounting packet transmission, in minutes. Ranging 0 to 32767. The default value is 0, namely, real-time accounting is disabled.

Description

Use the radius timer realtime-accounting command to configure the timeout value for transmitting real-time accounting packet to RADIUS server.

Use the undo radius timer realtime-accounting command to restore the default value.

After the user passed authentication, NAS transfers user's real-time accounting information to the RADIUS server every specified time. If the real-time accounting request fails, the RADIUS client will allow the user to continue using network service.

Example

# Configure the timeout value to two minutes for transmitting RADIUS real-time accounting packet.

[VG] radius timer realtime-accounting 2

5.1.32 radius timer response-timeout

Syntax

radius timer response-timeout seconds

undo radius timer response-timeout

View

System view

Parameter

seconds: Timeout value for the RADIUS server, in seconds. The value range 1 to 65535. The default value is 10 seconds.

Description

Use the radius timer response-timeout command to configure the timeout value of the RADIUS server.

Use the undo radius timer response-timeout command to restore the default value.

When replies are required for sent packets (like authenticating request packets), the timeout value should be configured, and packets will be retransmitted in case of timeout.

Example

# Configure the timeout value for the RADIUS server to 5 seconds.

[VG] radius timer response-timeout 5

5.1.33 redialtimes

Syntax

redialtimes redialtimes-number

View

Access service ID view

Parameter

redialtimes-number: Number of dial attempts that a subscriber is allowed to make at each dial stage, which is in the range of 1 to 10 and defaults to 3.

Description

Use the redialtimes command to configure the number of dial attempts that each subscriber is allowed to make at each dial stage.

In the case of calling number process, this configuration can be omitted.

As for the card number process, the dial order is access number, prepaid card number, PIN, and finally the destination number, given that the access number is correctly dialed. During the dial process, even a digit input mistake can cause the failure of the entire dial process. In order to give a chance for users to correct the errors, it is necessary to set a maximum number of dial attempts that the users are allowed to make at each dial stage.

& Note:

Before you set the maximum number of dial attempts, you should note that:

l This setting takes the same effect at each dial stage. For example, in a card number dial process, the attempts that you are allowed to enter a correct card number, PIN, and destination number are the same.

l The redialtimes command literally means the redial attempts. But it actually refers to the total dial times. Therefore, if you want to redial n times, you must make redialtimes-number equals n+1. For example, if you want to give three redial chances, set redialtimes-number to 4.

Related command: gw-access-number, process-config, card-digit, and password-digit.

Example

# Set the number of card number/PIN redial attempts for the access number 18901 to 4, that is, five dial attempts.

[VG-voice-dial-anum18901] redialtimes 5

5.1.34 reset voice aaa-client statistic

Syntax

reset voice aaa-client statistic

View

Any view

Parameter

None

Description

Use the reset voice aaa-client statistic command to clear the RADIUS Server related state statistics that are stored on the RADIUS Client (VG).

Related command: display voice aaa-client statistic.

Example

# Clear the statistics related to the state of the RADIUS Server.

[VG] reset voice aaa-client statistic

5.1.35 reset voice vcc

Syntax

reset voice vcc { all | call-record | statistics }

View

Any view

Parameter

all: Clears all the information.

call-record: Clears call records.

statistics: Clears the statistics.

Description

Use the reset voice vcc command to clear the VCC-related information.

You can clear all or part of the information related to the VCC module by information type.

Related command: display voice vcc and debugging voice vcc.

Example

# Clear the call records.

[VG] reset voice vcc call-record

5.1.36 selectlanguage

Syntax

selectlanguage { disable | enable }

View

Access service ID view

Parameter

disable: Disables language selection.

enable: Enables language selection.

Description

Use the selectlanguage command to enable or disable the voice gateway to play the prompt language selection message in the voice caller number process.

By default, language selection is disabled.

This command only applies to the voice caller number process (process-config voice-caller).

With language selection enabled, the voice gateway plays a message to ask users to select a prompt language and then input the called number as prompted. If language selection is disabled, the called number input prompt is played in Chinese after authentication.

Related command: process-config.

Example

# Enable prompt language selection.

[VG-voice-dial-anum600] process-config voice-caller

[VG-voice-dial-anum600] selectlanguage enable

5.1.37 server-authorization

Syntax

server-authorization

undo server-authorization

View

Voice AAA service

Parameter

None

Description

Use the server-authorization command to enable local users to request the RADIUS server for authorization.

Use the undo server-authorization command to disable local users from requesting the RADIUS server for authorization.

By default, this command is not used in local user configuration in voice AAA service view. Namely, local users are authenticated only locally, without requesting the RADIUS server for authorization. Non-local users are authenticated and authorized through the RADIUS server.

After this command is executed, local users will first pass local authentication and then request the RADIUS server for authorization, and the RADIUS server will respond with the authorization result. The information of voice RADIUS users needs to be configured on the RADIUS server, which performs authorization after authentication succeeds; otherwise the RADIUS will reject the authorization request.

A card user must use the car number and password generated by the RADIUS server.

By default, local users do not request the RADIUS server for authorization.

& Note:

If clienttype has been set to VSA-NOSTANDARD, you must use the server-authorization command to enable local users to request the RADIUS server for authorization. If clienttype has been specified with any of the other parameters, this command is optional.

Example

# Enable local users to request the RADIUS server for authorization.

[VG-voice-aaa] server-authorization

Chapter 6 GK Client Configuration Commands

6.1 GK Client Configuration Commands

6.1.1 area-id (gk-client)

Syntax

area-id string

undo area-id

View

Voice GK client view (gk-client)

Parameter

string: Area ID that contains 1 to 31 characters including number 0 to 9 and character #.

Description

Use the area-id command to configure the area ID of an H.323 GW.

Use the undo area-id command to delete the specified area ID.

By default, no area ID is configured for any H.323 GW.

The area-id is mainly used to facilitate the identifying of gateway type by the GK Server. The gateway and the GK Server reach an agreement on the gateway type in advance, for example, consider that the area-id 1# represents the voice gateway and that the area-id 2# represents the video gateway, etc. When the gateway communicates normally with the GK Server, the GK will judge the gateway type according to the area-id information sent by the gateway.

To set the H.323 area ID in the Voice GK client voice is mainly to facilitate the GK Server to identify the types of GK client. An agreement on the related types is reached beforehand between the GK client and the GK Server, e.g. area ID 1# represents a voice GK client, prefix 2# represents a video GK client, etc. When a normal communication is going on between the GK client and the GK Server, the GK judges the client type according to the area ID information received. Up to 30 area-ides can be configured in voice GK client view.

Configuring H.323 area IDs in voice GK client view will allow a GK Server to identify GK Client types.

Related command: match-template and entity.

Example

# Set the area ID of the CK Client to 6# and 1#.

[VG-voice-gk] area-id 6#

[VG-voice-gk] area-id 1#1

6.1.2 debugging voice ras

Syntax

debugging voice ras event

View

Any view

Parameter

event: Outputs the RAS message records.

Description

Use the debugging voice ras command to enable the debugging on the RAS messages between the GK Client and the GK Server.

Example

# Enable the debugging on the RAS messages between the GK Client and the GK Server.

[VG] debugging voice ras event

6.1.3 display voice gateway

Syntax

display voice gateway

View

All views

Parameter

None

Description

Use the display voice gateway command to display the state information about the registration of the GW with the GK Server.

Executing this command will display information including the registration state of the GW, GW alias, local telephone number list of the GW, etc.

& Note:

The display information of this command may vary with networks of different manufacturers’ gateways.

Example

# Enable support for displaying multiple GW-IDs.

[VG-1041-voice-gk]display voice gateway

GATEWAY ENDPOINT INFORMATION

============================

GW_STATE = Registered

Current GK_ID = pspublic

Config Master GK_ID = pspublic

Current GW information :

H323-ID gw01sub0

Current GK information :

H323-ID gw01sub0

GW_STATE = Registered

Current GK_ID = pspublic

Config Master GK_ID = pspublic

Current GW information :

H323-ID gw01sub1

Current GK information :

H323-ID gw01sub1

GW_STATE = Registered

Current GK_ID = pspublic

Config Master GK_ID = pspublic

Current GW information :

H323-ID gw01sub2

{

E164-ID 4001

}

Current GK information :

H323-ID gw01sub2

{

E164-ID 4001

}

GW_STATE = Registered

Current GK_ID = pspublic

Config Master GK_ID = pspublic

Current GW information :

H323-ID gw01sub3

{

E164-ID 4000

}

Current GK information :

H323-ID gw01sub3

{

E164-ID 4000

}

GW_STATE = Registered

Current GK_ID = pspublic

Config Master GK_ID = pspublic

Current GW information :

H323-ID gw01sub4

{

E164-ID 4487

}

Current GK information :

H323-ID gw01sub4

{

E164-ID 4487

}

Table 6-1 Enable support for displaying multiple GW-IDs

Item	Description
GW_STATE	State of the current GW terminal
Current GK_ID	ID of the currently registered GK
Config Master GK_ID	Display of master GK_ID
Config Slave GK_ID	Display of slave GK_ID
Current GW information : H323-ID	H323 ID of the current GW
E164 111	E164 number of the current GW
E164 112	Support for multiple numbers
Current GK Information:	Local information fed back by GK
NONE	None

# Enable support for displaying multiple H323-IDs.

GATEWAY ENDPOINT INFORMATION

============================

GW_STATE = Registered

Current GK_ID = pspublic

Config Master GK_ID = pspublic

Current GW information :

H323-ID gw01

{

E164-ID 4487

E164-ID 4000

E164-ID 4001

}

Current GK information :

H323-ID gw01

{

E164-ID 4487

E164-ID 4000

E164-ID 4001

}

6.1.4 gk-2nd-id

Syntax

gk-2nd-id gk-name gk-addr gk-ipaddress [ ras-port ]

undo gk-2nd-id

View

Voice GK client view (gk-client)

Parameter

See the parameter description for the gk-id command.

Description

Use the gk-2nd-id command to configure the name and IP address of the secondary GK Server for the GW.

Use the undo gk-2nd-id command to delete the name and IP address of the secondary GK Server for the GW.

In case there is communications problem (such as timeout) between the GK Client and the primary GK Server or the primary GK Server is unavailable, the VG can initiate its registration request to the secondary GK Server.

You can also configure the port information of the secondary GK Server by using this command.

& Note:

Before you can configure the name and address of the secondary GK Server, you must use the gk-id command to configure the name and address of the primary GK Server.

Related command: gk-id.

Example

# Adopt the GK Server named gk-backup at 1.1.1.2 as the secondary GK Server, and use the default RAS communication port.

[VG-voice-gk] gk-id gk-center gk-addr 1.1.1.1

[VG-voice-gk] gk-2nd-id gk-backup gk-addr 1.1.1.2

6.1.5 gk-client

Syntax

gk-client

View

Voice view

Parameter

None

Description

Using the gk-client command, you can enter Voice GK client view, and configure the GK parameters for voice. Using the quit command, you can exit this view.

For related commands, see area-id, gk-2nd-id, gk-id, gw-address, gw-id, ras-on.

Example

# Enter gk-client view.

[VG-voice] gk-client

[VG-voice-gk]

6.1.6 gk-id

Syntax

gk-id gk-name gk-addr gk-ipaddress [ ras-port ]

undo gk-id

View

Voice GK client view (gk-client)

Parameter

gk-name: GK Server name, which is a string comprising 1 to 128 case-sensitive characters.

gk-ipaddress: IP address of the GK Server.

ras-port: RAS communication port of the GK Server, which is an integer in the range of 1 to 65535 and defaults to 1719.

Description

Use the gk-id command to configure name and IP address of the primary GK Server for the GW.

Use the undo gk-id command to delete the name and IP address of the primary GK Server for the GW.

You can configure IP address, name and RAS communication port of the primary GK Server in the gk-id command. With the information, the GK Client can find an appropriate primary GK Server and register with it.

Related command: area-id, gw-id, gk-2nd-id, gw-address, and ras-on.

Example

# Adopt the GK Server named gk-center at 1.1.1.1 as the primary GK Server, and use the default RAS communication port.

[VG-voice-gk] gk-id gk-center gk-addr 1.1.1.1

6.1.7 gk-security call

Syntax

gk-security call enable

gk-security call disable

View

Voice GK view (gk-client)

Parameter

enable: Enable the security call.

disable: Disable the security call.

Description

Use the gk-security call enable command to set the GK Client to enable security call.

Use the gk-security call disable command to disable the GK Client to disable the security call.

By default, security call is disabled.

After configure the GK client to enable the security call, the calling GW obtains the token ring from the GK Client in a call and transmitted in a pass-through mode to the call GW. The GW then sends the token ring to the GK Server.

In some networks where the called GK Server cannot handle the token ring, you must disable the GK to use the security call.

Example

# Disable the GK Client to use the security call.

[VG-voice-gk] gk-security call disable

6.1.8 gk-security register-pwd

Syntax

gk-security register-pwd { cipher | simple } password

undo gk-security register-pwd

View

Voice GK view (gk-client)

Parameter

cipher: Uses the echoed password in cipher text..

simple: Uses the echoed password in simple text.

password: The set password within from 1 to 16 printable characters except spaces.

Description

Use the gk-security register-pwd command to configure the GK register password.

Use the undo gk-security register-pwd command to cancel the GK register password.

By default, GK Client has no register password.

Once you configure the register password on the GK Client, all the registration needs this password.

Example

# Configure the GK register password to comware in cipher text.

[VG-voice-gk] gk-security register-pwd cipher comware

6.1.9 gw-address

Syntax

gw-address { dial-bundle-number dial-number | ethernet interface-number | ip address }

undo gw-address

View

Voice GK client view (gk-client)

Parameter

dial-bundle-number: Source IP address bound with one PPPoE port. The IP address can be obtained manually or dynamically.

dial-number: PPPoE port number.

ethernet: Source IP address bound with an Ethernet port. The IP address can be obtained manually or dynamically. If both primary and secondary IP addresses are configured for the interface, the primary one will be used as the GW source IP address.

interface-number: Ethernet interface number.

ip: Source IP address bound with a static IP address.

address: IP address you have entered.

Description

Use the gw-address command to configure the source IP address of the VG.

Use the undo gw-address command to delete the source IP address of the VG.

By default, no source IP address is configured for the VG.

Caution:

l The bound GW IP address and the IP addresses of peer H.323 entities (Gatekeeper, Terminal, MCU and so on) should be mutually accessible. Otherwise call failures will occur.

l Only the VG 10-40 and VG 10-41 voice gateway support the PPPoE Client function, that is, support the dial-bundle-number parameter.

Related command: area-id, gk-2nd-id, gk-id, gw-address, and ras-on.

Example

# Set the source IP address of the VG to 1.1.1.1.

[VG-voice-gk] gw-address ethernet 0

6.1.10 gw-id

Syntax

gw-id namestring

undo gw-id [ namestring ]

View

Voice GK client view (gk-client)

Parameter

namestring: GW alias (GW identification) that is a string comprising 1 to 128 case-sensitive characters.

Description

Use the gw-id command to configure an alias for the GW.

Use the undo gw-id command to delete the GW alias.

By default, no GW alias is configured, that is, GW alias is null.

An alias is configured for the VG for the purpose of identifying it and registered it with the GK Server. A GW can have only one alias and the new alias will replace the old one.

Related command: area-id, gk-2nd-id, gk-id, gw-address, and ras-on.

& Note:

After switching between the non-gw-id mode and multi-gwid mode, the configured gw-id features will be lost

Example

# Assign the alias hangzhou-gw to the GW.

[VG-voice-gk] gw-id hangzhou-gw

6.1.11 multi-gwid

Syntax

multi-gwid

undo multi-gwid

View

Voice GK view (gk-client)

Parameter

None

Description

Use the multi-gwid command to enable multi GW-ID mode.

Use the undo multi-gwid command to disable GW-ID mode.

By default, multi GW-ID mode is disabled.

When you have enabled the multi-gwid function, a parameter will be added to the gw-id command to specify the analog port for configuration. At this point, you have to configure different gw-id for each analog port, making one GW to register with GK as several GW. The VG 80-20 does not support multi GW-ID mode.

& Note:

In the switching between the single gw-id mode and multiple gw-id mode, the original gw-id configuration does not exist.

Example

# Configure the multi GW-ID mode on the voice GW with four analog ports.

[VG-voice-gk] multi-gwid

[VG-voice-gk] gw-id 0 gw01

[VG-voice-gk] gw-id 1 gw02

[VG-voice-gk] gw-id 2 gw03

[VG-voice-gk] gw-id 3 gw04

6.1.12 ras-on

Syntax

ras-on

undo ras-on

View

Voice GK client view (gk-client)

Parameter

None

Description

Use the ras-on command to enable the GK Client function.

Use the undo ras-on command to disable the GK Client function.

By default, the GK Client function is disabled.

Only after the GK Client is enabled can the VG and the GK Server maintain normal communications. If the function is not enabled, the VG will be unable to set up a connection with the GK Server.

Example

# Enable the GK Client function.

[VG-voice-gk] ras-on

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