Syntax
address { ip
ip-address | ras }
undo address { ip | ras }
View
Voice entity view
Parameter
ip ip-address:
Indicate a VoIP dial entity session destination, i.e. the called IP
address.
ras: Router
uses RAS recommendation 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 is
configured.
This command is used to configure the
network address for the VolP voice entity. The system supports the following
two VolP routing policies at present.
l
Static routing policy: Find the IP address of
destination voice gateway in static mode according to address ip ip-address
command.
l
Dynamic routing policy: The router and GK Server
interact with RAS information after the address ras command is
configured. GK will dynamically send back the peer voice gateway address that matches
the called number to the router.
Related command: address sip, match-template.
Example
# Configure the destination IP address
corresponding to the called 12345 as 10.1.1.2.
[H3C-voice-dial-entity1]
match-template 12345
[H3C-voice-dial-entity1] address ip 10.1.1.2
Syntax
area {
north-america | custom | europe }
undo area
View
Voice subscriber-line view
Parameter
north-america: Busy tone type of the switch connected to this subscriber line is
of North America standard.
custom: Busy
tone type of the switch connected to this subscriber line is defined by the
users.
europe: Busy
tone type of the switch 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, and it can only perform configuration to the first
voice subscriber line on the voice card. If successful, the configuration will
be effective for all the voice subscriber lines of the voice card.
When this subscriber line is connected to a
common user line of a program-controlled switch, if the user on the switch side
hooks on first, only by detecting the busy tone can the router know the user
on-hooking operation. Since different switches execute different prompt tone
schemes, there exist different frequency spectrum characteristics. This command
is used to set the frequency spectrum characteristic used by the router 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 0/0/1.
[H3C-voice-line0/0/1] area
north-america
Syntax
area-id string
undo area-id
View
Voice entity view
Parameter
string: Area
ID, an integer in the range of 0 to 9. The “#” can be used.
Description
Use the area-id command to configure
the area ID of voice GW.
Use the undo area-id command to
cancel the specified area ID.
By default, no area ID of voice GW 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, entity.
Example
# Configure the VoIP voice entity 101 with
the area ID 6#.
[H3C-voice-dial-entity101] area-id 6#
Syntax
busytone-t-th time-threshold
undo busytone-t-th
View
FXO voice subscriber line view
Parameter
time-threshold: Threshold of busy tone detection. It ranges from 2 to 12, with a
bigger value meaning longer detection time. The threshold defaults to 2, that
is, the device hangs up upon two contiguous detections of busy tone.
Description
Use the busytone-t-th command to
configure the threshold of busy tone detection.
Use the undo busytone-t-th command
to restore the default.
The actual busy tone data does not always match
the configured parameter. If the difference is large, detection inaccuracy may
occur, resulting in on-hook failure or improper on-hook. You can however, tune
the threshold of busy tone detection to achieve detection accuracy. For
example, you can eliminate improper on-hooks caused by busy tone data
inaccuracy by increasing the time for busy tone detection.
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 threshold of busy tone detection
to 3.
[H3C-voice-line0/0/1] busytone-t-th 3
Syntax
caller-permit calling-string
undo caller-permit { calling-string | all }
View
Voice entity view
Parameter
all: All
callers.
calling-string: 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
“0123456789ABCD#*.!+%[]()-”. The meanings of the characters are
described in the following table:
Table 1-1 Meanings of the characters in string
|
Character
|
Meaning
|
|
0-9
|
Numbers from 0 to 9. Each means a
digit.
|
|
ABCD
|
Each character means a digit.
|
|
# and *
|
Each means a
valid digit.
|
|
.
|
A wildcard. It can match any digit of
a valid number. For example, 555. . . . matches any
string that begins with 555 and with four additional characters.
|
|
!
|
The character or characters right in
front of it does not appear or appears once. For example, 56!1234 can match
51234 and 561234.
|
|
+
|
The character or characters right in front of it appears once or several times. But its appearance at the beginning of the
whole number means the number is E.164-compliant.
|
|
-
|
Hyphen. It connects two values (the
smaller one before it and the bigger one after it) to indicate a range. For
example, “1-9” means numbers from 1 to 9 (inclusive).
|
|
%
|
The character or characters right in front of it does not appear, or appears
once or several times.
|
|
[ ]
|
Selects one character from the group.
For example, [1-36A] can match only one character
among 1, 2, 3, 6, and A.
|
|
( )
|
A group of characters. For example, (123A) means a string “123A”. It is usually used with
“!”, “%”, and “+”. For example,
“408(12)+” can match 40812 or 408121212.
But it cannot match 408. That is, “12” can appear continuously
and it must at least appear once.
|
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
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 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.
[H3C-voice-dial-entity2] caller-permit
660268$
[H3C-voice-dial-entity2] caller-permit
$
# Configure voice entity 2 to permit the
calling numbers beginning with 20 to call in.
[H3C-voice-dial-entity2] caller-permit
20
Syntax
cid display
undo cid display
View
Voice subscriber line view
Parameter
None
Description
Use the cid display command
to enable caller identification display.
Use the undo cid display
command to disable caller identification display.
By default, caller identification display
is enabled.
This command is applicable to FXS subscriber
lines.
When functioning as the called, the FXS
module can send caller identification information to its called phone between
the first and second rings. When disabled to send caller identification
information, the FXS interface sends the character “P” received
from the IP side instead. Thus, the called phone is unable to display caller identification
information.
Example
# Enable caller identification display on
voice subscriber line 1/0/0.
[H3C-voice-line1/0/0] cid display
Syntax
cid
enable
undo cid
enable
View
Voice subscriber
line view
Parameter
None
Description
Use the cid enable command to enable
CID on the FXO interface.
Use the undo cid enable command to
disable CID on the FXO interface.
By default, CID is enabled on the FXO
interface.
This command applies to FXO voice subscriber
lines only.
With CID enabled, the FXO interface can
receive the modulated caller identification data from an analog line between
the first ring and second rings and then send the data demodulated with FSK to
the IP side.
With CID disabled, the local FXO interface
does the following when the calling party sends a calling number:
l
If a number is configured in the match template
for the POTS entity associated with the local FXO interface, the interface
substitutes this number for the calling number and sends it to the called side.
l
If wildcard dots (.) are used in the number
configured in the match template for the POTS entity associated with the local
FXO interface, the interface substitutes zeros for the calling number’s
digits in the place of dots, for example, 1000 for 1… and then sends the
substitution number to the called side.
Example
# Enable CID on FXO voice subscriber line
1/0/0.
[H3C-voice-line1/0/0] cid enable
Syntax
cid send
undo cid send
View
Voice subscriber line view
Parameter
None
Description
Use the cid send command to enable
the FXO or FXS module to send calling numbers to the IP side.
Use the undo cid send command to
disable the FXO or FXS module 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 subscriber
lines only.
After you configure the undo cid send
command, the FXO interface does not send any number to the called side,
regardless of whether the calling party has sent a calling number and
regardless of whether a number is configured in the match template for the
voice entity associated with the FXO interface.
Example
# Enable the FXO voice subscriber line
3/0/0 to send calling numbers to the IP side.
[H3C-voice-line3/0/0] cid send
# Disable the FXS voice subscriber line
1/0/0 to send calling numbers to the IP side.
[H3C-voice-line1/0/0] undo cid send
Syntax
cid type { complex
| simple }
View
Voice subscriber line view
Parameter
complex: Caller
identification information is transmitted in multiple-data message format
(MDMF).
simple: Caller
identification information is transmitted in single-data message format (SDMF).
Description
Use the cid
type command to configure the format of transmitted information about the
calling party.
Two formats are available: multiple data
message format (MDMF) and single data message format (SDMF). When the remote
end supports one format only, you must use the same setting at the local end.
This command applies to both FXO and FXS subscriber
lines.
Example
# Set the format of the transmitted caller
identification information to SDMF on voice subscriber line 1/0/0.
[H3C-voice-line1/0/0] cid type simple
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 FXO, FXS,
E&M subscriber lines and digital E1 voice subscriber line. 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 1/0/0.
[H3C-voice-line1/0/0] undo cng-on
Syntax
compression {
1st-level | 2nd-level | 3rd-level | 4th-level } { g711alaw
| g711ulaw | g723r53 | g723r63 | g729a | g729ab |
g729r8 | g729br8 | g726r16 | g726r24 | g726r32 |
g726r40 | clear-channel }
undo compression { 1st-level | 2nd-level | 3rd-level | 4th-level
}
View
Voice entity view
Parameter
1st-level:
Indicates the first selected voice compression method.
2nd-level:
Indicates the second selected voice compression method.
3rd-level:
Indicates the third selected voice compression method.
4th-level:
Indicates the fourth selected voice compression method.
g711alaw:
Specifies G.711 A-law codec (defining the pulse code modulation technology),
requiring the bandwidth of 64 kbps, usually adopted by Europe.
g711ulaw:
Specifies G.711μ-law codec, requiring the bandwidth of 64 kbps, usually
adopted in the North America and Japan.
g723r53:
Specifies G.723.1 Annex A codec, requiring the bandwidth of 5.3 kbps.
g723r63:
Specifies G.723.1 Annex A codec, requiring the bandwidth of 6.3 kbps.
g729a:
Specifies G.729 Annex A codec (a simplified version of G.729 codec), requiring
the bandwidth of 8 kbps.
g729r8:
Specifies G.729 (the voice coding technology using conjugate
algebraic-code-excited linear-prediction) codec, requiring the bandwidth of 8 kbps.
g729ab: Adds
G.729 Annex B codec on the basis of G.729a.
g729br8: Specifies
G.729 Annex B codec, requiring the bandwidth of 8 kbps. This codec supports voice
activity detection (VAD), discontinuous transmission (DTX), and comfort noise
(CNG).
g726r16:
Specifies G.726 Annex A codec. It uses the adaptive
differential pulse code modulation (ADPCM) technology, requiring the bandwidth
of 16 kbps.
g726r24:
Specifies G.726 Annex A codec. It uses ADPCM, requiring
the bandwidth of 24 kbps.
g726r32:
Specifies G.726 Annex A codec. It uses ADPCM, requiring
the bandwidth of 32 kbps.
g726r40:
Specifies G.726 Annex A codec. It uses ADPCM, requiring
the bandwidth of 40 kbps.
clear-channel: Specifies the G.Clear codec, which is used for transparent data
transmission with a bandwidth of 64 Kbps (nodata compression).
Description
Use the compression command to
configure the voice codec according to priority level.
Use the undo compression command to
restore the default value.
By default, g729r8 codec is set.
g711alaw and
g711ulaw coding provide high-quality voice transmission, while requiring
greater bandwidth.
g726r16, g726r24, g726r32, and g726r40 are
widely adopted now for voice coding. They use the ADPCM technology and provide
multiple bandwidth options.
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 codec 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.<
