Académique Documents
Professionnel Documents
Culture Documents
V200R002C01
01
Date
2012-04-20
Notice
The purchased products, services and features are stipulated by the contract made between Huawei and the
customer. All or part of the products, services and features described in this document may not be within the
purchase scope or the usage scope. Unless otherwise specified in the contract, all statements, information,
and recommendations in this document are provided "AS IS" without warranties, guarantees or representations
of any kind, either express or implied.
The information in this document is subject to change without notice. Every effort has been made in the
preparation of this document to ensure accuracy of the contents, but all statements, information, and
recommendations in this document do not constitute the warranty of any kind, express or implied.
Website:
http://www.huawei.com
Email:
support@huawei.com
Issue 01 (2012-04-20)
Commissioning engineers
Symbol Conventions
The symbols that may be found in this document are defined as follows.
Symbol
Description
DANGER
WARNING
CAUTION
Issue 01 (2012-04-20)
TIP
NOTE
ii
Command Conventions
The command conventions that may be found in this document are defined as follows.
Convention
Description
Boldface
Italic
[]
{ x | y | ... }
[ x | y | ... ]
{ x | y | ... }*
[ x | y | ... ]*
&<1-n>
Change History
Changes between document issues are cumulative. Therefore, the latest document issue contains
all updates made in previous issues.
Issue 01 (2012-04-20)
iii
Contents
Contents
About This Document.....................................................................................................................ii
1 SIPAG Configuration...................................................................................................................1
1.1 SIP AG Overview...............................................................................................................................................2
1.2 SIP AG Features Supported by the AR2200......................................................................................................2
1.3 Configuring the AR2200 to Work in SIP AG Mode..........................................................................................6
1.4 Configuring a SIP AG Interface.........................................................................................................................7
1.4.1 Establishing the Configuration Task.........................................................................................................7
1.4.2 Configuring Media and Signaling IP Address Pools.................................................................................8
1.4.3 Setting Parameters on the SIP AG Interface.............................................................................................8
1.4.4 Checking the Configuration.....................................................................................................................15
1.5 (Optional) Setting Parameters for Voice Interfaces..........................................................................................16
1.5.1 Setting Parameters for an FXS Interface.................................................................................................16
1.5.2 Setting parameters for a BRA Interface...................................................................................................18
1.5.3 Setting Parameters for a PRA Interface...................................................................................................19
1.6 (Optional) Setting Voice Parameters on a SIP AG...........................................................................................21
1.6.1 Setting System Parameters......................................................................................................................21
1.6.2 Setting SIP Stack Parameters..................................................................................................................30
1.6.3 Setting DSP Parameters...........................................................................................................................32
1.7 Configuring a SIPAG User...............................................................................................................................43
1.7.1 Establishing the Configuration Task.......................................................................................................44
1.7.2 Setting Parameters for a SIP AG User.....................................................................................................44
1.7.3 Configuring a SIP AG User Group..........................................................................................................46
1.7.4 (Optional) Configuring Functions or Services in a SIP Service Data Profile.........................................47
1.7.5 Checking the Configuration.....................................................................................................................49
1.8 (Optional) Configuring the BEST Service.......................................................................................................50
1.9 Resetting a SIP AG...........................................................................................................................................52
1.10 Maintaining a SIP AG....................................................................................................................................53
1.10.1 Clearing SIP AG Statistics....................................................................................................................53
1.11 Configuration Examples.................................................................................................................................53
1.11.1 Example for Configuring a SIP AG......................................................................................................53
1.11.2 Using a PRA Trunk to Connect the SIP AG to a PBX..........................................................................57
1.11.3 Example for Configuring the BEST Service.........................................................................................61
2 PBX Configuration......................................................................................................................66
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iv
Contents
Contents
Issue 01 (2012-04-20)
vi
1 SIPAG Configuration
SIPAG Configuration
Issue 01 (2012-04-20)
1 SIPAG Configuration
VoIP
On the Public Switched Telephone (PSTN), telephone lines are occupied exclusively. The fees
for toll calls are high.
To reduce data and voice fees and meet service requirements, voice over IP (VoIP) is used.
VoIP transmits voice services over the IP network. In VoIP, a voice gateway encapsulates voice
signals into frames and transmits them as IP packets. Currently, IP phone gateways implement
communication between the PSTN network and IP network. As PC-to-phone, phone-to-PC, and
phone-to-phone technologies have developed, the voice quality is improved greatly. VoIP can
meet commercial requirements.
Voice Gateway
The IP network is a packet switched network. The transmission costs on the IP network are lower
than those on the PSTN; therefore, VoIP will take place traditional voice phones gradually.
Replacing all phone networks with VoIP will take high costs. The step-by-step migration
solution is recommended.
In most enterprise voice solutions, phone networks of branches are reserved and the IP network
is used between branches.
In the step-by-step migration solution, a voice gateway is required to connect the two types of
networks. A router is often used as the voice gateway.
IMS
The IP Multimedia Core Network Subsystem (IMS) is an architectural framework for providing
IP multimedia services, including audio, video, text, and instant messages. It was designed by
the wireless standards body 3rd Generation Partnership Project (3GPP) in Release 5.
SIP
The Session Initiation Protocol (SIP) is a text-based signaling protocol. SIP messages are
classified into request and response messages. SIP can be used for creating, modifying, and
terminating two-party or multiparty sessions. SIP can be used for multimedia conferences,
remote education, and Internet calls.
SIPAG
A SIP access gateway (SIP AG) is a voice gateway that exchanges SIP signals with other devices
between the PSTN/ISDN and IP network. It can implement VoIP functions.
1 SIPAG Configuration
A user connects to the SIP AG through the PSTN. The SIP AG converts analog signals into
digital signals and compresses and packetizes the digital signals so that the voice packets
can be transmitted on the IP network.
Voice packets are transmitted to the SIP AG through the IP network. The IP voice gateway
of the called party converts the voice packets into analog voice signals and transmits them
to the terminal of the called party through the PSTN.
NOTE
l The AR2200 series routers support the voice features only after a DSP module is installed.
l To provide voice services for POTS users, 4FXS1FXO board is required.
l To provide voice services for ISDN users, 2BST board is required.
IMS/IP
SIPAG
POTS
Eth1/0/0
Modem
FAX
A user picks up the phone and the SIP AG detects offhook signals.
2.
3.
The user hears the dial tone played by the session application of the SIP AG and starts to
dial up.
4.
The session application of the SIP AG collects the number dialed by the user.
5.
The session application of the SIP AG matches the number with preconfigured called
number profiles.
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1 SIPAG Configuration
6.
When the voice gateway of the calling party successfully matches the number with a
preconfigured called number profile, the number is mapped to the voice gateway of the
called party.
7.
The SIP AG initiates a call over the IMS and establishes a logical channel for each call.
Then the channel is used to send and receive voice data.
8.
The IMS searches for the destination phone and initiates a call.
9.
After the called party picks up the phone, a call is set up. After the calling party or called
party hangs up the phone, the call is ended.
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Service Type
Introduction
Configure
d on the
SIP AG or
Not
Yes
Three-party service
Yes
Call waiting
service
Yes
MWI service
Yes
Malicious call
identification
(MCID) service
The user that registers the MCID service with the carrier
can query the phone number of the attacker that initiates
malicious calls after performing relevant operations.
Yes
Call transfer
service
Yes
Call conference
service
Yes
1 SIPAG Configuration
Service Type
Introduction
Configure
d on the
SIP AG or
Not
Calling line
identification
presentation
(CLIP) service
No
Calling line
identification
restriction (CLIR)
service
No
Distinctive ringing
service
No
Differentiated
ringback tone
service
No
Advice of charge
(AoC) service
No
Polarity reversal
charging service
No
Polarity reversal
pulse charging
service
No
No
Completion of
Calls to Busy
Subscriber (CCBS)
service
No
Multiple MSN
numbers on a
POTS interface
No
Hotline service
No
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1 SIPAG Configuration
Service Type
Introduction
Configure
d on the
SIP AG or
Not
Anonymous call
service
No
License Support
The SIPAG function is used with a license. To use the SIPAG function, apply for and purchase
the following license from the Huawei local office:
AR2200 value-added service package for voice services
To use the BEST function, you must purchase the CM&BEST License in addition to the precedng
licenses.
Applicable Environment
You can configure the AR2200 to work in SIP AG or PBX mode. Before configuring SIP AG
service features, configure the AR2200 to work in SIP AG mode. You can run the display voice
service-mode command to view the working mode of the AR2200. If the AR2200 works in
PBX mode, delete the PBX configurations and configure the AR2200 to work in SIP AG mode.
If the AR2200 works in SIP AG mode, skip this configuration.
Pre-configuration Tasks
Before configuring the AR2200 to work in SIP AG mode, complete the following task:
l
Procedure
Step 1 Run:
system-view
1 SIPAG Configuration
Step 4 Run:
quit
After the AR2200 is configured to work in SIP AG mode, restart the AR2200 to make the configuration
take effect.
----End
Applicable Environment
To allow the SIP AG and IMS network to exchange media and signaling streams, the media and
signaling IP addresses, signaling port number, and transmission protocol need to be configured.
Pre-configuration Tasks
Before configuring a SIP AG interface, complete the following task:
l
Data Preparation
To configure a SIP AG interface, you need the following data.
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1 SIPAG Configuration
No.
Data
SIP AG interface number, media and signaling IP addresses, signaling port number,
IP addresses and port numbers of primary and secondary proxy servers, transmission
protocol, and home domain name
Context
A SIP AG interface must obtain media and signaling IP addresses from media and signaling IP
address pools respectively. The signaling IP address pool stores IP addresses of SIP AG
interfaces and the media IP address pool stores IP addresses of media streams. Media and
signaling IP address pools can contain the same IP addresses.
Procedure
Step 1 Run:
system-view
1 SIPAG Configuration
Context
To allow the SIP AG and IMS network to exchange media and signaling streams, set the media
and signaling IP addresses, signaling port number, and transmission protocol.
Procedure
Step 1 Run:
system-view
Command
Remarks
Media IP address
The signaling IP
address of the SIP AG
interface must be
obtained from the
signaling IP address
pool.
By default, no
signaling port number
is configured for a SIP
AG interface.
Paramet
ers for
the
primary
proxy
server
Issue 01 (2012-04-20)
Static IP
address
By default, no static IP
address is configured
for the primary proxy
server.
Parameter
1 SIPAG Configuration
Command
Remarks
DNS-A
domain name
and port
number
By default, no DNS-A
domain name or port
number is configured
for the primary proxy
server.
DNS-NAPTR
domain name
DNS-SRV
domain name
Transmission protocol
transfer transfer-protocol
By default, a SIP AG
interface uses the User
Datagram Protocol
(UDP) protocol as the
transmission protocol.
home-domain home-domain-value
By default, no home
domain name is
configured for a SIP
AG interface.
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Parameter
Command
Remarks
Description
description description
By default, no
description is
configured for a SIP
AG interface.
Profile index
profile profile-index
Domain name
ag-domain ag-domain-name
By default, no domain
name is configured for
a SIP AG interface.
conference-factory-uri uri
By default, no
conference factory
URI is configured for a
SIP AG interface.
Authent
ication
By default, the
authentication mode is
user.
Authenticatio
n mode
10
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1 SIPAG Configuration
Parameter
Command
Remarks
paramet
ers
By default, no
authentication user
name or password is
configured for a SIP
AG user.
Timer value
DTMF parameters
dtmf-transmission-mode
{ thoroughly | erase }
Fax/
Modem
paramet
ers
Fax/Modem
codec
negotiation
mode
Packetization
interval in
transparent
transmission
mode
By default, the
packetization interval
in transparent
transmission mode is
10 ms.
VBD attribute
type
VBD codec
mode
VBD payload
type
Fax
transmission
mode
Modem
transmission
mode
Authenticatio
n user name
and password
11
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1 SIPAG Configuration
Parameter
Command
Remarks
Paramet
ers of
the
seconda
ry proxy
server
Static IP
address and
port number
By default, no static IP
address or port number
is configured for the
secondary proxy
server.
DNS-A
domain name
and port
number
By default, no DNS-A
domain name or port
number is configured
for the secondary proxy
server.
DNS-NAPTR
domain name
secondary-proxy-addr dns-naptr
dns-naptr-domain-name
DNS-SRV
domain name
proxy-dhcp-option option-value
Software parameters
Different software
parameters may use
different value ranges.
For details, see ag
number-parameter.
String parameters
ag string-parameter strpara-name
strpara-value
RFC
2833
transmis
sion
RFC 2833
DTMF
transmission
RFC 2833
hookflash
transmission
12
Parameter
Mode in
which RFC
2833
transmission
is started
based on
negotiation
Proxy
function
s
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1 SIPAG Configuration
Command
Remarks
nte-negotiation-mode mode-value
By default, a SIP AG
uses the probe mode to
detect a proxy server.
Mode in
which a SIP
AG is dual
homed to
proxy servers
By default, a SIP AG
supports dual homing
but does not support
automatic switchover.
Mode used by
a SIP AG to
update the
proxy server
address
proxy-refresh-mode { no-switch |
defer | immediate }
By default, a SIP AG
updates the proxy
server IP address after
a delay (defer mode).
Registration of a SIP AG
with the specified proxy
server
proxy-switchover { primary |
secondary }
To maintain or
troubleshoot the
current proxy server,
run this command.
RFC
2198
redunda
ncy
transmis
sion
Mode in
which RFC
2198
redundancy
transmission
is started
redundancy-negotiation-mode
negotiation-mode
Transmission
of RFC 2833
audio data
using RFC
2198
redundancy
transmission
By default, a SIP AG is
disabled from using
RFC 2198 redundancy
transmission to
transmit the RFC 2833
audio data.
Mode in
which RFC
2198
redundancy
transmission
is started
redundancy-start-mode mode-value
13
Parameter
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1 SIPAG Configuration
Command
Remarks
VBD
transmission
using RFC
2198
redundancy
transmission
By default, a SIP AG is
enabled to transmit
VBD using RFC 2198
redundancy
transmission.
Voice
transmission
using RFC
2198
redundancy
transmission
redundancy-voice { enable |
disable }
By default, a SIP AG
does not use RFC 2198
redundancy
transmission to
transmit voice.
Registrar URI
register-server-uri uri
By default, no registrar
URI is configured for a
SIP AG.
Ringing parameters
By default, ringing
parameters of a SIP AG
is empty.
sdp-negotiation-mode { local |
remote }
service-logic service-logic-index
14
1 SIPAG Configuration
Parameter
Command
Remarks
MWI
subscrip
tion
function
MWI
subscription
function
Register
status
subscription
function
UA profile
subscription
function
By default, the UA
profile subscription
function is enabled on a
SIP AG.
user-defined-profile codec-priority
pri-value codec-value pt-value
user-defined-profile service-priority
pri-value srvid-value
----End
Run the display voice sipag [ sipag-interface-id { running | config } ] command to check
the configuration of SIP AG parameters.
Run the display voice voip-address command to check the signaling and media IP
addresses pool.
----End
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1 SIPAG Configuration
Applicable Environment
An FXS interface connects to a POTS phone. To achieve high transmission efficiency on an
FXS interface, properly set parameters for the FXS interface on the AR2200, including physical
attributes, electrical attributes, and KC attributes.
Pre-configuration Tasks
Before setting parameters for an FXS interface, complete the following task:
l
Data Preparation
To set parameters for an FXS interface, you need the following data.
No.
Data
Polarity reversal pulse level width, polarity reversal mode, and dialing mode
High-level pulse width, low-level pulse width, KC accounting mode, and voltage
operating
Procedure
Step 1 Run:
system-view
16
1 SIPAG Configuration
| soft }
17
1 SIPAG Configuration
kc voltage voltage
Applicable Environment
A basic rate access (BRA) interface connects to an ISDN phone. On the AR2200, you can enable
the BRA interface Layer 2 monitoring, remote power supply, automatic deactivation, and alarm
functions, and set the working mode and Layer 1 activation mode on a BRA interface.
Pre-configuration Tasks
Before setting parameters for a BRA interface, complete the following task:
l
Data Preparation
To set parameters for a BRA interface, you need the following data.
No.
Data
Interface working mode, automatic deactivation delay, and Layer 1 activation mode
Procedure
Step 1 Run:
system-view
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1 SIPAG Configuration
Applicable Environment
A PRA interface connects to a PBX or PSTN network. On the AR2200, you can enable the
CRC4 check, E1 interface Layer 2 monitoring, and E1 interface pulse code modulation (PCM)
alarm functions, and set the CRC alarm threshold and E1 interface signaling mode on a PRA
interface.
Pre-configuration Tasks
Before setting parameters for a PRA interface, complete the following task:
l
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1 SIPAG Configuration
Data Preparation
To set parameters for a PRA interface, you need the following data.
No.
Data
Procedure
Step 1 Run:
system-view
e1t1 e1-voice
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20
1 SIPAG Configuration
Applicable Environment
The AR2200 working in SIP AG mode can exchange information with a softswitch device using
SIP. Different countries and regions use different voice parameter standards; therefore, set voice
parameters on the SIP AG in accordance with local standards.
Pre-configuration Tasks
Before setting system parameters, complete the following task:
l
Data Preparation
To set system parameters, you need the following data.
Issue 01 (2012-04-20)
No.
Data
Country/Region identifier
MWI mode
AC amplitude of the ringing current, frequency of the ringing current, and cadence
ratio
Huawei Proprietary and Confidential
Copyright Huawei Technologies Co., Ltd.
21
No.
Data
1 SIPAG Configuration
l Interval between the time when the ACK message is received and the time when
the FSK is transmitted
l Maximum duration between the time when the DT-AS is transmitted and the time
when the ACK message is received
l Duration of the DT-AS
l Level of the DT-AS
l Level of the FSK
l Number of bits of the FSK synchronization mask
7
Applicable Environment
A country/region identifier is configured on a SIP AG so that user terminals connect to the SIP
AG can comply with the local standard.
Procedure
Step 1 Run:
system-view
22
1 SIPAG Configuration
Context
Hookflash or flash is a button on a telephone that simulates quickly hanging up and then picking
up again (a quick off-hook/on-hook/off-hook cycle). The hookflash can be pressed by a calling
party or a called party:
l
Hookflash pressed by a called party: If the called party user A wants to transfer an incoming
call to user B, user A can press the hookflash and dial the number of user B.
Hookflash pressed by a calling party: User A calls user B. User B answers the call and talks
with user A. User A can press the hookflash and dial the number of user C after hearing a
special dial tone.
If user C is busy, user A can press the hookflash and talk with user B.
If user C does not respond for a long period of time, user A can press the hookflash and
talk with user B.
If the phone of user C rings, user A hangs up and user B hears the ringback tone. User C
picks up the phone and talks with user B.
Whether a called party can be transferred to a toll call is restricted by the outgoing right of
the called party.
Procedure
Step 1 Run:
system-view
23
1 SIPAG Configuration
NOTE
The lower threshold for hookflash pressing must be 50 ms less than the upper threshold for hookflash
pressing.
----End
Context
If there are leave messages, the user device configured with the MWI function makes the
indicator on or plays a tone, indicating that there are leave messages. You can set the MWI mode
according to user habits.
Procedure
Step 1 Run:
system-view
Context
G.711, also known as Pulse Code Modulation (PCM), is a commonly used waveform codec. G.
711 defines two main compression algorithms, the -law algorithm (used in North America &
Japan) and A-law algorithm (used in Europe and China). A-law encoding takes a 13-bit signed
linear audio sample as input. -law encoding takes a 15-bit signed linear audio sample as input.
Procedure
Step 1 Run:
system-view
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1 SIPAG Configuration
Procedure
Step 1 Run:
system-view
Run:
sctp checksum receive enable
Run:
sctp checksum send enable
Context
Different countries and regions use different ringing standards. You can set the AC amplitude
of the ringing current to adjust the ringing tone volume, voice pitch, cadence ratio, and initial
ringing function on the AR2200 to meet local standards.
Procedure
Step 1 Run:
system-view
25
1 SIPAG Configuration
Step 2 Run:
voice
Run:
ring frequency { 16hz | 25hz | 50hz }
Run:
user-defined-ring ring-index { first-ring first-ring-period | first-interval
first-interval | second-ring second-ring-period | second-interval secondinterval | third-ring third-ring-period | third-interval third-interval }*
Run:
stop-initial-ring { enable | disable }
Context
When the CLIP service is registered, CLIP parameters in offhook state need to be configured
on the AR2200 so that the AR2200 can work with the phone terminal. Generally, default
parameter settings are used. If CLIP parameters are not set properly, change relevant CLIP
parameters.
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26
1 SIPAG Configuration
Procedure
Step 1 Run:
system-view
Run:
clip offhook ack-fsk-interval ack-fsk-interval
The interval between the time when the ACK message is received and the time when the
frequency-shift keying (FSK) is transmitted in offhook state is set.
l
Run:
clip offhook dtas-ack-interval dtas-ack-interval
The maximum duration between the time when the dual tone-alerting signal (DT-AS) is
transmitted and the time when the ACK message is received in offhook state is set.
l
Run:
clip offhook dtas-duration dtas-dur-value
The duration of the dual tone-alerting signal (DT-AS) in offhook state is set.
l
Run:
clip offhook dtas-level dtas-level
Run:
clip offhook fsk-level fsk-level
Run:
clip offhook mark-signal-bit mark-signal-bit
The number of bits of the FSK synchronization mask in offhook state is set.
----End
Context
When the CLIP service is registered, CLIP parameters in onhook state need to be configured on
the AR2200 so that the AR2200 can work with the phone terminal. Generally, default parameter
settings are used. If CLIP parameters are not set properly, change relevant CLIP parameters.
Procedure
Step 1 Run:
system-view
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1 SIPAG Configuration
Run:
clip onhook channel-seize-bit channel-seize-bit
Run:
clip onhook dtas-duration dtas-dur-value
Run:
clip onhook dtas-fsk-interval dtas-fsk-interval
The interval between the time when the DT-AS is transmitted and the time when the FSK
is transmitted in onhook state is set.
l
Run:
clip onhook dtas-level dtas-level
Run:
clip onhook fsk-level fsk-level
Run:
clip onhook mark-signal-bit marksignal-bit
The number of bits of the FSK synchronization mask in onhook state is set.
----End
Context
The AR2200 provides uplink bandwidth control. When the system detects that the uplink
bandwidth usage reaches the configured upper threshold, it restricts calls and generates an alarm.
If the uplink bandwidth is insufficient, the system processes calls based on user levels. Common
users may not obtain services.
Procedure
Step 1 Run:
system-view
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1 SIPAG Configuration
Run:
media-bandwidth-control enable
The AR2200 is enabled to restrict calls when the uplink bandwidth is enabled.
l
Run:
media-bandwidth-control maximum max-bandwidth
Run:
media-bandwidth-control reserved-for-emergency reserved-bandwidth
The reserved bandwidth for emergency calls must be smaller than the maximum uplink bandwidth
configured by the media-bandwidth-control maximum command.
----End
Procedure
l
Run the display voice configuration command to check the voice configuration.
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1 SIPAG Configuration
Run the display voice sip-reg-count-per-second command to check the number of SIP
Register messages initiated per second.
----End
Applicable Environment
SIP is an IETF-defined signaling protocol widely used for controlling communication sessions
such as voice and video calls over Internet Protocol (IP). SIP, RTP, RTCP, RTSP, and other
protocols constitute a SIP protocol stack.
Pre-configuration Tasks
Before setting SIP protocol stack parameters, complete the following task:
l
Data Preparation
To set SIP protocol stack parameters, you need the following data.
No.
Data
Procedure
Step 1 Run:
system-view
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1 SIPAG Configuration
Run:
entity-based-sessions-timer enable
Run:
min-se min-seperiod
Run:
session-progress session-progressperiod
Run:
session-rel-progress session-rel-progressperiod
Run:
t1 t1period
Run:
t2 t2period
Run:
t4 t4period
Run:
td tdperiod
Procedure
Step 1 Run:
system-view
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1 SIPAG Configuration
Run:
header-folding enable
Run:
field-header max-forwards max-forwards-value
Run:
field-header organization organization-head
Run:
field-header server server-head
Run:
field-header user-agent user-agent-head
Procedure
l
----End
32
1 SIPAG Configuration
Applicable Environment
The digital signal processing (DSP) collects, converts, filters, measures, enhances, compresses,
or identifies signals and coverts the signal from an analog to a digital form.
The DSP module converts analog voice signals into digital signals and stores a certain number
of digital signals into packets for transmission. To improve the voice communication quality,
the DSP needs to further process voice signals.
Pre-configuration Tasks
Before setting DSP parameters, complete the following task:
l
Data Preparation
To set DSP parameters, you need the following data.
No.
Data
Default DSP channel code type and default interval at which the DSP channel
packetizes RTP packets
T.30 redundancy parameter value of the T.38 fax, T.4 redundancy parameter value
of the T.38 fax, fax training mode, and maximum fax training rate
Alarm threshold of the dynamic jitter buffer, initial value of the dynamic jitter buffer,
maximum value of the dynamic jitter buffer, maximum value of the static jitter buffer,
minimum value of the dynamic jitter buffer, minimum value of the static jitter buffer,
and initial value of the static jitter buffer of a DSP channel
RTP payload type value, G.726-16k payload type value, G.726-24k payload type
value, G.726-32k payload type value, G.726-40k payload type value, NTE payload
type value, redundancy payload type value, and VBD payload value of a DSP channel
Interval at which a DSP channel sends RTCP packets and threshold for the number
of severe degrade seconds
Data event transmission mode, special process, DTMF transmission mode, echo
cancellation function, input gain, output gain, jitter buffer mode, NLP mode, and DSP
working mode in a DSP template
33
1 SIPAG Configuration
Context
A user may hear the user's echo in the phone receiver in a conversation. If a proper delay in the
transmitted or received signal is set, the echo can be removed. If the delay exceeds 25 ms, the
voice quality deteriorates and the conversation ends. You can enable echo cancellation on a DSP
channel to remove echoes.
Procedure
Step 1 Run:
system-view
Context
PLC is a technique that masks the effects of packet loss in VoIP communications. PLC is
effective only when the packet loss ratio is low. During communication, the average packet loss
ratio may be low, but a high burst packet loss ratio results in severe voice quality deterioration.
PLC can insert a static frame in the place where a packet is lost, regenerate a packet received
prior to the lost one, or generate an analog voice packet. If packets are lost during communication
and PLC is not used, the voice communication is interrupted. You can use a proper PLC
algorithm to minimize effects of packet losses.
Procedure
Step 1 Run:
system-view
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1 SIPAG Configuration
Context
To save network bandwidth, enable silence compression on a DSP channel. When no voice is
detected, the encoder generates short silence codes, but does not generate voice compression
codes. In addition, the encoder notifies the receiver of silence start until the voice is restored.
The silence compression function reduces the number of sent voice packets.
Procedure
Step 1 Run:
system-view
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----End
Setting the Default DSP Channel Code Type and the Default Interval at Which a
DSP Channel Packetizes RTP Packets
This section describes how to set the default DSP channel code type and the default interval at
which a DSP channel packetizes RTP packets.
Context
The voice encoding technique encodes pulse-code modulation (PCM) samples into bits (frames).
This technique ensures robustness of voice services when the error code, network jitter, or burst
traffic occurs on a link. At the receiver side, voice frames are encoded into PCM samples, and
then are converted into voice waveforms. Different voice encoding techniques provide different
voice quality and a good voice quality requires high bandwidth.
In VoIP, before voice data are transmitted as UDP packets, the Real-time Transport Protocol
(RTP) processes the voice data. RTP is used to transmit real-time data and can transmit audio
and video data.
Procedure
Step 1 Run:
system-view
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1 SIPAG Configuration
The default DSP channel code type and the default interval at which a DSP channel packetizes
RTP packets are set.
----End
Procedure
Step 1 Run:
system-view
Run:
fax redundancy-t4 redundancy-t4value
Run:
fax redundancy-t30 redundancy-t30value
Run:
fax training-mode { e2e | local }
Run:
fax training-rate { v17 | v29 | v27 }
Run:
fax v8negotiate enable
V8 negotiation is enabled.
----End
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1 SIPAG Configuration
Context
Delay variations in voice packet arrival time can occur because of network congestion or route
changes. To reduce sound distortion caused by the delay jitter and packet loss, a jitter buffer is
used. You can set proper jitter buffer parameters to minimize delay variations so that packets
can be processed in a timely manner and smooth voice communication can be provided as much
as possible.
Procedure
Step 1 Run:
system-view
Run:
jitter-buffer adapt-jb-threshold adapt-jb-threshold
Run:
jitter-buffer init-adapt-jb init-adapt-jb-value
Run:
jitter-buffer init-fixed-jb normal-fixed-jb-value
Run:
jitter-buffer max-adapt-jb max-adapt-jb-value
Run:
jitter-buffer min-adapt-jb min-adapt-jb-value
Run:
jitter-buffer max-fixed-jb max-fixed-jb-value
Run:
jitter-buffer min-fixed-jb min-fixed-jb-value
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1 SIPAG Configuration
Procedure
Step 1 Run:
system-view
Run:
payload-type clear-mode clearmode-value
Run:
payload-type g726-16k g726-16k-value
Run:
payload-type g726-24k g726-24k-value
Run:
payload-type g726-32k g726-32k-value
Run:
payload-type g726-40k g726-40k-value
Run:
payload-type nte nte-value
Run:
payload-type redundancy redundancy-value
Run:
payload-type vbd vbd-value
39
1 SIPAG Configuration
Context
RTCP monitors the quality of service and conveys information about participants in an on-going
session. RTCP periodically sends packets to all the participants in the session to monitor the
quality of service and obtain identity information about the participants.
Procedure
Step 1 Run:
system-view
Run:
rtcp rtcp-interval rtcp-interval
Run:
rtcp rtcpxr enable
The RTP Control Protocol Extended Reports (RTCP XR) function is enabled.
l
Run:
rtcp sev-degradethreshold sev-degradethresholdval
Run:
rtcp vqm enable
Context
DSP resources are limited and users have different requirements for DSP resources. To control
and allocate DSP resources properly, set the DSP resource control mode and the resource
threshold in hierarchical control mode.
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1 SIPAG Configuration
Procedure
Step 1 Run:
system-view
Run:
resource-threshold mode { priority | normal }
Run:
resource-threshold { threshold1 threshold1 | threshold2 threshold2 |
threshold3 threshold3 }
Context
To customize DSP parameters for data services, configure a DSP template. After a DSP template
is configured, specify the template for users according to the port and phone number. The DSP
template improves the call connection rate. After a template is specified successfully, parameters
in the DSP template take effect immediately.
Procedure
Step 1 Run:
system-view
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Run:
data-event data-event-value
Run:
dsp-special-flow dsp-special-flow-value
Run:
dtmf dtmf-value
Run:
echo-cancel { enable | disable }
Run:
input-gain input-gain-value
Run:
output-gain output-gain-value
Run:
jitter-buffer { dynamic | static }
Run:
nlp nlp-value
Run:
work-mode work-mode-value
Run:
vbd redundancy
Context
You can enable a digital signal processor (DSP) channel to work in loopback mode, and set the
loopback mode (PCM-side loopback test and IP-side loopback test). When the DSP channel
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between the calling party and called party cannot transmit signals or can transmit signals only
in one direction, run the loop-back command to locate the fault. If the calling party hears the
echo in a PCM-side loopback test, the speech channel between the calling phone and the calling
DSP channel is functioning properly. If the called party hears the echo in an IP-side loopback
test, the speech channel between the called phone and the calling DSP channel is functioning
properly.
To control resources of DSP channels, prohibit the DSP channels. The prohibited DSP channels
cannot participate in resource allocation.
Procedure
Step 1 Run:
system-view
Run:
loop-back loopback-type channel
Run:
prohibit channel [ count ]
Procedure
l
Run the display voice dsp-attribute command to check the DSP configuration.
Run the display voice dsp-template command to check the DSP template configuration.
----End
43
1 SIPAG Configuration
Applicable Environment
On the IMS, a SIP AG is directly connected to a user terminal. You need to set parameters for
users on the SIP AG so that the users can use services on the IMS.
Pre-configuration Tasks
Before configuring a SIP AG user, complete the following tasks:
l
Data Preparation
To configure a SIP AG user, you need the following data.
No.
Data
SIP AG user port number, SIP AG ID for the SIP AG user, and phone number of the
SIP AG user
(Optional) SIP AG user in the SIP service data profile, and phone number of the SIP
AG user in the SIP service data profile
Applicable Environment
On the IMS, a SIP AG is directly connected to a user terminal. You need to set parameters for
users on the SIP AG so that the users can use services on the IMS.
Procedure
Step 1 Run:
system-view
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1 SIPAG Configuration
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Parameter
Command
Default Setting
Timeslot status
NOTE
This command is valid only on
VE1 interfaces.
Maximum
number of
automatic
recovery
attempts from
deterioration
auto-resume-limit auto-resume-limit
20
SIP AG's
capability to
support the
BELL ANS
flag
Calling
number format
Sequence in
which the
calling number
of a SIP AG
user is
displayed
clip-transmission-sequence { after-ring |
before-ring }
after-ring
Power-off
interval
dc-time dc-time
10 ms
Single-tone
ANSbar signal
detection
detect-ansbar-by-single-tone { enable |
disable }
disable
dsp-input-gain dsp-input-gain
0 dB
NOTE
This command is valid only on VE1 and BRA
interfaces.
disable
NOTE
This command is valid only on
FXS interfaces.
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1 SIPAG Configuration
Parameter
Command
Default Setting
DSP chip
output gain
dsp-output-gain dsp-output-gain
0 dB
DSP template
name
dsp-template dsp-template
Service
termination of
a specified SIP
AG user
Extended
phone number
extend-telno telno-value
[ sipagusergroup usergroup-id ]
FSK call
display mode
BELL_202
FSK delay
fsk-time fsk-time
10 ms
User type
DEL
SIP AG user
priority
cat3
TAS mode of
CLIP
NO-TAS
UNI fault
reporting
function
disable
NOTE
This command is valid only on VE1 and BRA
interfaces.
Voice quality
enhancement
(VQE)
function
disable
Target VQE
level
vqe-agc-level vqe-agc-level
13
VQE noise
suppression
disable
Noise
suppression
level
vqe-sns-level vqe-sns-level
12 dB
----End
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1 SIPAG Configuration
Procedure
Step 1 Run:
system-view
Run:
group-name usergroup-name
Run:
auth username auth-username password { cipher | ha1 { cipher | simple authpassword1 } | simple auth-password2 }
Run:
precinct-mode { local | remote }
The mode used to manage users in the SIP AP user group is configured.
l
Run:
register-uri-mode { inneruser | alone }
Run:
subscribe ua-profile { enable
| disable
Run:
uri uri
A URI is configured.
l
Run:
endservice
47
1 SIPAG Configuration
Applicable Environment
After a SIP AG user is configured, communication can be implemented. You can enable other
services for the SIP AG user according to user requirements.
Before configuring other services for a SIP AG user, run the service-right conf disable
command in the SIP AG service data profile to disable the call conference service.
Procedure
Step 1 Run:
system-view
Issue 01 (2012-04-20)
Function/Service
Command
Default Setting
Authentication user
name and password
centrex-dial { directly |
secondary }
Direct dialing
Centrex number
centrex-number centrexnumber
Empty
Centrex prefix
centrex-prefix centrex-prefix
CFNR forward-to
number and timeout
interval
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1 SIPAG Configuration
Function/Service
Command
Default Setting
Hotline number
service-right anonymous-call
enable
disable
service-right call-diversion
enable
disable
enable
service-right call-transfer
enable
disable
enable
service-right cw enable
enable
Hotline service
disable
Malicious call
identification (MCID)
service
disable
MWI service
disable
Terminal portability
service
disable
Three-party service
service-right three-party
enable
enable
mwi-mode { deferred |
immediate | combine }
UUS1 service
disable
UUS2 service
disable
UUS3 service
disable
----End
Run the display voice sipaguser [ sipaguser-name ] command to check the configuration
of the SIP AG user.
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----End
Applicable Environment
Many enterprises or organizations deploy IP phones in their branches and use SIP servers in
headquarters to control calls of remote branches in a centralized manner. When communication
between a branch and the headquarters fails, the call service and other voice services in the
branch are interrupted. The BEST service can be configured to solve this problem. When
communication between a branch SIP AG and the headquarters SIP server fails, the SIP AG in
the branch starts to manage local calls to ensure uninterrupted voice services in the branch. When
communication between the branch and headquarters is restored, the headquarters SIP server
controls all calls.
As shown in Figure 1-2, an AR router functions as a SIP AG and connects to an IMS network.
When communication with the IMS network fails, the router manages local calls.
Figure 1-2 BEST networking
IMS
Network
Eth1/0/0
port 2/0/0
SIPAG
port 2/0/2
port 2/0/1
POTS
IP Phone
FAX
CAUTION
After enabling the BEST service, you must restart the router for the service to take effect, which
will interrupt all services on the router. Therefore, confirm your operation before enabling the
BEST service.
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1 SIPAG Configuration
Pre-configuration Tasks
Before configuring the BEST service, complete the following tasks:
l
Configuring a SIP AG
NOTE
By default, a SIP AG uses the probe mode to detect a proxy server. The BEST service cannot be used
in probe mode; therefore, after creating a SIP AG, run the proxy-detect-mode command to set the
proxy detection mode to option or register.
The SIP AG bound to the BEST service must have user phone numbers configured.
Procedure
1.
Run:
system-view
Run:
voice
Run:
best enable
Run:
quit
Run:
quit
Run:
save
Run:
reboot
The BEST service takes effect only after you save the configuration and restart the router.
8.
l When configuring the BEST service, the call prefix is bound to the enterprise default and DN
set defaultdialplan configured by using the enterprise default dn-set defaultdialplan
command.
9.
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1 SIPAG Configuration
Run the display voice best config command to check the BEST service configuration.
Run the display voice best probe command to check the SIP AG bound to the BEST service.
Run the display voice best status command to check the status of the BEST service.
Applicable Environment
To make modified SIP AG parameters take effect, reset the SIP AG.
Pre-configuration Tasks
Before resetting a SIP AG, complete the following task:
l
Procedure
Step 1 Run:
system-view
CAUTION
Exercise caution when you run this command because resetting a SIP AG interrupts running
services.
----End
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1 SIPAG Configuration
CAUTION
The cleared SIP AG statistics cannot be restored. Exercise caution when you run reset
commands.
Procedure
Step 1 Run the reset sctp-association-statistics command in the SIP AG view to clear statistics about
SCTP associations on a SIP AG.
Step 2 Run the reset sctp-global-statistics command in the user view to clear global SCTP statistics.
----End
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1 SIPAG Configuration
IMS
Network
GE1/0/0
port 2/0/0
SIPAG
port 1/0/0
port 2/0/1
POTS
POTS
ISDN
Configuration Roadmap
The configuration roadmap is as follows:
1.
2.
Create SIP AG interface and set parameters for the SIP AG interface.
3.
Data Preparation
To complete the configuration, you need the following data:
l
Interface through which the SIP AG exchanges media and signaling streams with the IMS
network: GigabitEthernet1/0/0
SIP AG's ports directly connected to user terminals: port 2/0/0, port 2/0/1, and port 1/0/0
Phone numbers of user terminals connected to port 2/0/0, port 2/0/1, and port 1/0/0:
11111111, 11112222, and 11113333
Procedure
Step 1 Configure the AR2200 to work in SIP AG mode and reboot the device.
<Huawei> system-view
[Huawei] voice
[Huawei-voice] service-mode sipag
The modification takes effect only after you save the data and reboot the devic
e. Are you sure to change the protocol configuration?(y/n)[n]:
y
[Huawei-voice] quit
[Huawei] quit
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<Huawei> save
The current configuration will be written to the device.
Are you sure to continue? (y/n)[n]:y
It will take several minutes to save configuration file, please wait..........
..............
Configuration file had been saved successfully
Note: The configuration file will take effect after being activated
<Huawei> reboot
Info: The system is comparing the configuration, please wait.
System will reboot! Continue ? [y/n]:y
Info: system is rebooting ,please wait...
# Set the IP address pool for exchanging media and signaling streams.
[Huawei] voice
[Huawei-voice] voip-address signalling interface gigabitethernet 1/0/0 1.1.1.1
[Huawei-voice] voip-address media interface gigabitethernet 1/0/0 1.1.1.1
port 2/0/0
agid 1
base-telno 11111111
quit
port 2/0/1
agid 1
base-telno 11112222
quit
port 1/0/0
agid 1
base-telno 11113333
quit
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1 SIPAG Configuration
:
:
:
:
:
:
:
:
:
:
:
:
:
:
:
:
:
:
:
:
:
:
:
:
:
:
:
:
:
:
:
:
1
1.1.1.1
5060
1.1.1.1
UDP
2.2.2.2
255.255.255.255
255.255.255.255
255.255.255.255
5060
65535
IP
huawei.com
1: Default
0: Default
0: None
Enable
Disable
Disable
Remote
Manual switch over
Probe
Run the display voice sipaguser [ sipaguser-name ] command to view detailed information
about the SIP AG users.
<Huawei> display voice sipaguser 1
Slotid/Subcard/Portid
: 2/0/0
AGID
:
Base telno
: 11111111
GroupID
:
----End
Configuration Files
Configuration file of the Router
#
voice
voip-address signalling interface GigabitEthernet 1/0/0
1.1.1.1
voip-address media interface GigabitEthernet 1/0/0
1.1.1.1
#
sipag 1
signalling-addr 1.1.1.1
5060
media-addr 1.1.1.1
primary-proxy-addr static 2.2.2.2
5060
home-domain huawei.com
#
sipaguser 1 port 2/0/0
base-telno 11111111
agid 1
#
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port 2/0/1
11112222
port 1/0/0
11113333
#
interface GigabitEthernet1/0/0
ip address 1.1.1.1 255.255.255.0
#
return
IMS
Network
GE1/0/0
Router
port 1/0/0
PRA trunk
PBX
POTS A
POTS B
Configuration Roadmap
The configuration roadmap is as follows:
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1 SIPAG Configuration
1.
2.
3.
Create SIP AG interface and set parameters for the SIP AG interface on the Router.
4.
Create SIP AG users and set parameters of SIP AG users on the Router.
5.
Configure the trunk type on the PBX as PBX and use the PBX as the user-side device.
Data Preparation
To complete the configuration, you need the following data:
l
Interface through which the Router exchanges media and signaling streams with the IMS
network: GigabitEthernet1/0/0
Procedure
Step 1 Configure the AR2200 to work in SIP AG mode on the Router and reboot the device.
<Huawei> system-view
[Huawei] sysname Router
[Router] voice
[Router-voice] service-mode sipag
The modification takes effect only after you save the data and reboot the devic
e. Are you sure to change the protocol configuration?(y/n)[n]:
y
[Router-voice] quit
[Router] quit
<Huawei> save
The current configuration will be written to the device.
Are you sure to continue? (y/n)[n]:y
It will take several minutes to save configuration file, please wait..........
..............
Configuration file had been saved successfully
Note: The configuration file will take effect after being activated
<Router> reboot
Info: The system is comparing the configuration, please wait.
System will reboot! Continue ? [y/n]:y
Info: system is rebooting ,please wait...
# Set the IP address pool for exchanging media and signaling streams.
[Router] voice
[Router-voice] voip-address signalling interface gigabitethernet 1/0/0 1.1.1.1
[Router-voice] voip-address media interface gigabitethernet 1/0/0 1.1.1.1
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1 SIPAG Configuration
port 1/0/0
agid 1
extend-telno 11111111
extend-telno 11112222
quit
Step 5 Configure the trunk type on the PBX as PBX and use the PBX as the user-side device.
The configurations on different types of PBXs are different, so the configuration procedure is
not mentioned here. If the AR is used as the PBX, see 2.12 Configuring a Trunk Group.
Step 6 Verify the configuration.
Then user terminals on the SIP AG can implement voice communication.
Run the display voice sipag [ sipag-interface-id { running | config } ] command to view detailed
information about the SIP AG.
<Router> display voice sipag 1 config
AGID
Dynamic signalling IP address name
Signalling IP
Signalling port
Dynamic media IP address name
Media IP
Transfer mode
Primary proxy IP 1
Primary proxy IP 2
Secondary proxy IP 1
Secondary proxy IP 2
Primary proxy port
Secondary proxy port
Primary proxy domain name
Secondary proxy domain name
Proxy address mode
Home domain name
SIP profile index
Service logic index
Server Address DHCP option
Description
AG domain name
Phone context
Register URI
Conference factory URI
Subscribe to UA profile
Subscribe to reg state
Subscribe to MWI
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:
:
:
:
:
:
:
:
:
:
:
:
:
:
:
:
:
:
:
:
:
:
:
:
:
:
:
:
1
1.1.1.1
5060
1.1.1.1
UDP
2.2.2.2
255.255.255.255
255.255.255.255
255.255.255.255
5060
65535
IP
huawei.com
1: Default
0: Default
0: None
Enable
Disable
Disable
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1 SIPAG Configuration
Run the display voice sipaguser [ sipaguser-name ] command to view detailed information
about the SIP AG users.
<Router> display voice sipaguser 1
Slotid/Subcard/Portid
: 1/0/0
AGID
: 1
Base telno
:
GroupID
:
Extend telno
: 11111111
GroupID
:
Extend telno
: 11112222
GroupID
:
Priority
: cat3
Uri report
: Disable
Auto limit
: 20
B channel 0
: normal
B channel 1
: normal
B channel 2
: normal
B channel 3
: normal
B channel 4
: normal
B channel 5
: normal
B channel 6
: normal
B channel 7
: normal
B channel 8
: normal
B channel 9
: normal
B channel 10
: normal
B channel 11
: normal
B channel 12
: normal
B channel 13
: normal
B channel 14
: normal
B channel 15
: normal
B channel 16
: normal
B channel 17
: normal
B channel 18
: normal
B channel 19
: normal
B channel 20
: normal
B channel 21
: normal
B channel 22
: normal
B channel 23
: normal
B channel 24
: normal
B channel 25
: normal
B channel 26
: normal
B channel 27
: normal
B channel 28
: normal
B channel 29
: normal
B channel 30
: normal
B channel 31
: normal
----End
Configuration Files
Configuration file of the Router
#
board add 0/1 1VE1-MFT
#
set workmode slot 2 e1t1 e1-voice
#
interface GigabitEthernet1/0/0
ip address 1.1.1.1 255.255.255.0
#
voice
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Networking Requirements
As shown in Figure 1-5, the Router functions as a voice gateway and connects to an IMS
network. To ensure that the Router can manage local calls when communication with the IMS
network fails, configure the BEST service on the Router.
Figure 1-5 BEST networking
IMS
Network
GE1/0/0
port 2/0/0
SIPAG
port 2/0/2
Router
POTS
1000
POTS
1001
Configuration Roadmap
The configuration roadmap is as follows:
Issue 01 (2012-04-20)
61
1.
2.
3.
4.
5.
6.
7.
1 SIPAG Configuration
Data Preparation
To complete the configuration, you need the following data:
l
Interface through which the SIP AG exchanges media and signaling streams with the IMS
network: Ethernet1/0/0
IP address and port number of the primary proxy server for the SIP AG: 10.10.1.205, 5066
SIP AG's ports directly connected to user terminals: port2/0/0 and port2/0/1
Phone numbers of user terminals directly connected to port2/0/0 and port2/0/1 of the SIP
AG: 1000 and 1001
Procedure
Step 1 Configure the AR2200 to work in SIP AG mode.
<Huawei> system-view
[Huawei] voice
[Huawei-voice] service-mode sipag
The modification takes effect only after you save the data and reboot the device.
Are you sure to change the protocol configuration?(y/n)[n]:
y
[Huawei-voice] quit
[Huawei] quit
<Huawei> save
The current configuration will be written to the device.
Are you sure to continue? (y/n)[n]:y
It will take several minutes to save configuration file, please wait..........
..............
Configuration file had been saved successfully
Note: The configuration file will take effect after being activated
<Huawei> reboot
Info: The system is comparing the configuration, please wait.
System will reboot! Continue ? [y/n]:y
Info: system is rebooting ,please wait...
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port 2/0/0
agid 1
base-telno 1000
quit
port 2/0/1
agid 1
base-telno 1001
quit
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Run the display voice best probe command to check the SIP AG bound to the BEST service.
The following information is displayed:
<Huawei> display voice best probe
The BEST probe mgid 1.
Run the display voice best status command to view the status of the BEST service. If the link
between the Router and the IMS network fails, the following information is displayed:
<Huawei> display voice best status
The BEST is running.
----End
Configuration Files
Configuration file of the Router
#
voice
voip-address signalling interface Ethernet 1/0/0 10.10.1.171
voip-address media interface Ethernet 1/0/0 10.10.1.171
best probe 1
#
sipserver
signalling-address ip 10.10.1.171 port 5060
media-ip 10.10.1.171
register-uri huawei.com
home-domain huawei.com
#
callprefix 1
enterprise default dn-set defaultdialplan
prefix 1
call-type category basic-service attribute 0
digit-length 4 6
#
sipag 1
signalling-addr 10.10.1.171 5061
media-addr 10.10.1.171
primary-proxy-addr static 10.10.1.205 5066
home-domain huawei.com
proxy-detect-mode option
#
sipaguser 1000 port 2/0/0
#
sipaguser 1001 port 1/0/2
base-telno 1001
agid 1
#
sipaguser 10000 port 1/0/1
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base-telno 1000
agid 1
#
return
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PBX Configuration
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Introduction
A traditional private branch exchange (PBX) manages incoming and outgoing calls of an
enterprise. It connects the enterprise to the Public Switched Telephone Network (PSTN) and
provides services for devices such as telephones, fax machines, and modems. It allows users in
the enterprise to call each other using extension phones and routes inter-office calls to the PSTN
through a trunk line.
Traditional PBXs cannot meet requirements for computer telephony integration (CTI) and voice
over IP (VoIP). In addition, these PBXs are expensive and do not use standard and open
platforms, bringing difficulties in interconnection between PBXs of different vendors. IP PBXs
overcome the limitations of traditional PBXs. IP PBXs are based on the IP protocol and provide
the local exchange and IP user access functions. IP PBXs integrate the voice communications
system of an enterprise into the enterprise's data network so that the enterprise can build a uniform
voice and data network connecting branches offices and staff around the world.
The AR2200 can function as a PBX to provide traditional PBX functions and IP PBX functions.
NOTE
l The AR2200 series routers support the voice features only after a DSP module is installed.
l To provide voice services for POTS users, 4FXS/1FXO board is required.
l To provide voice services for ISDN users, 2BST board is required.
Terms
l
DN set
A dial number (DN) set defines a group of numbers that are processed in the same way.
A DN set, a country code, and an area code identify the home area of a user; a DN set and
a call prefix determine the dialing plan for a user. DN sets divide a physical network or a
device into multiple logical networks.
Call prefix
A call prefix, an important attribute of the call service, defines a call number rule and
describes the call number distribution and routing plans in an exchange office. A call prefix
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identifies the service attribute (basic service or supplementary service; intra-office call,
national toll call, or international toll call) of a dialing plan and determines the range of
dialed number length. Call prefixes can also be used to control call permissions.
A PBX checks validity of a dialed number and connects or rejects the call based on the call
prefix. A call can be connected correctly only if the call matches the correct call prefix.
Therefore, proper call prefix configuration is the key to the call service. A call must match
at least one call prefix.
l
Call route
A call route binds an inter-office call prefix to a trunk group so that calls with the specified
call prefix are transmitted on the specified trunk line.
IMS
The IP Multimedia Core Network Subsystem (IMS) is an architectural framework for
providing IP multimedia services, including audio, video, text, and instant messages. It was
designed by the wireless standards body 3rd Generation Partnership Project (3GPP) in
Release 5.
SIP
SIP is an IETF-defined signaling protocol and runs at the control layer of the IMS. As an
application layer protocol, SIP establishes, modifies, or terminates multimedia sessions and
works with protocols such as the Real-time Transport Protocol (RTP), Real-Time Transport
Control Protocol (RTCP), Session Description Protocol (SDP), Real-time Stream Protocol
(RTSP), Domain Name System (DNS), Stream Control Transmission Protocol (SCTP),
and Transmission Control Protocol (TCP) to complete session setup and media negotiation.
H323
H.323 is a recommendation from the ITU Telecommunication Standardization Sector
(ITU-T) that defines the protocols to provide audio-visual communication sessions on
computer networks. H.323 initiates and terminates multimedia sessions, and can
dynamically change session attributes, including the required bandwidth, media type,
media encoding format, and support for broadcast.
PSTN user
PSTN users are Plain Old Telephone Service (POTS) users.
SIP UE
SIP user equipment (SIP UE) is a user device that connects to a PBX using the SIP protocol,
for example, an IP phone or a software terminal. SIP UEs connect to a PBX through the IP
network and obtain services from the PBX after registering on the PBX.
SIP Server
If SIP users need to communicate with each other through the AR2200, configure a SIP server
on the AR2200.
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SBC Proxy
If employees on a business trip use private network IP addresses, the SIPUEs cannot properly
connect to the PBX at the enterprise's headquarters. If employees within the enterprise use private
network IP addresses, they cannot call an external number through the SIP trunk. To solve this
problem, the SBC proxy function needs to be configured on AR2200.
CDR Server
Call detail records (CDRs) are details about calls recorded in real time. You can use a third-party
tool to analyze CDRs of calls and obtain call fees of users. To record and view CDRs of users,
specify a CDR server on the AR2200.
PRA trunk
A PRA trunk is connected to a VE1 interface and is usually used to connect two PBXs.
AT0 trunk
An AT0 trunk is connected to a foreign exchange office (FXO) interface and is used only
to connect to the upstream PSTN network.
SIP trunk
A SIP trunk is established on an IP link and connects a PBX and an IMS. There are two
types of SIP trunks: SIP AT0 trunk and common SIP trunk.
E1R2 trunk
The E1R2 trunk is bound to VE1 interfaces and is usually used to connect two PBXs. The
E1R2 trunk can be connected to a peer device using R2 signaling.
H323 trunk
An H323 trunk is established on an IP link. It connects a PBX and an IMS or connects
PBXs in different areas.
Trunks for inter-office calls with the same attributes are added to a trunk group. A trunk group
is associated with a call route to establish call sessions between exchange offices.
Call Route
A call route is bound to a trunk group to determine the trunk line for a specified call prefix. To
allow users to make inter-office calls, configure call routes.
Individual services
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Service Type
Description
Abbreviated dialing
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Service Type
Description
Call forwarding
Number barring
Do-not-disturb
Remote office
Secretary
Wake-up service
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Group services
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Service Type
Description
Call interception
Distinctive ringing
Enterprise RBT
Number change
Co-group pickup
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Service Type
Description
Simultaneous ringing
Sequential ringing
License Support
The PBX function is used with a license. To use the PBX function, apply for and purchase the
following license from Huawei local office:
Table 2-1 PBX licenses
License Name
Depends on
Remarks
This license has the following functions:
CM&BEST License
Value-added
service package
for voice services
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CT (Call Trunk)
License
Value-added
service package
for voice services
IVR (Interactive
Voice Response)
License
CM&BEST
License or CT
(Call Trunk)
license
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NOTE
l The PBX function is controlled by the CM&BEST, CT, and IVR licenses. For the functions provided
by each license, see the Remarks column in the preceding table.
l For the dependent licenses of the PBX function, see the Depends on column in the preceding table.
For example, the CM&BEST License depends on the value-added service package for voice services;
therefore, to use the CM function, load the value-added service package for voice services first.
l If a function has multiple capacity licenses, select one or multiple licenses. Multiple licenses can be
used together.
PBX
FXS1/0/0
User A
FXS1/0/1
SIPUE
User B
Figure 2-2 shows the configuration flowchart for intra-office calls. After mandatory
configurations are complete, users can call each other.
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Configure global
PBX parameters
Configure an
enterprise
Configure a call
prefix
Configure PBX
users
Configure a SIP
server
Configure a CDR
server
Configure
individual services
Configure group
services
End
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Branch
User B
AR
Enterprise A
SIP
PSTN
IP
SIP
AT0
AR
Headquarters
PBX
Enterprise A
User A
Figure 2-4 shows the configuration flowchart for inter-office calls. After mandatory
configurations are complete, users in the headquarters and branch can call each other.
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Configure global
PBX parameters
Configure an
enterprise
Configure a call
prefix
Configure PBX
users
Configure a trunk
group
Configure call
routes
Configure a SIP
server
Configure SBC
proxy
Configure a CDR
server
Configure
individual services
Configure group
services
End
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Applicable Environment
An FXS interface connects to a POTS phone. To achieve high transmission efficiency on an
FXS interface, properly set parameters for the FXS interface on the AR2200, including physical
attributes, electrical attributes, and KC attributes.
Pre-configuration Tasks
Before setting parameters for an FXS interface, complete the following task:
l
Data Preparation
To set parameters for an FXS interface, you need the following data.
No.
Data
Polarity reversal pulse level width, polarity reversal mode, and dialing mode
High-level pulse width, low-level pulse width, KC accounting mode, and voltage
operating
Procedure
Step 1 Run:
system-view
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| soft }
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kc voltage voltage
Applicable Environment
An FXO interface connects to a PSTN. To achieve high transmission efficiency on an FXO
interface, properly set parameters for the FXO interface on the AR2200, including the gain,
impedance, ring current, and feed.
Pre-configuration Tasks
Before setting parameters for an FXO interface, complete the following task:
l
Data Preparation
To set parameters for an FXO interface, you need the following data.
No.
Data
Procedure
Step 1 Run:
system-view
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If the RBT played for the caller through the FXO port is too loud, the FXO port may fail to detect the busy
tone from the caller; therefore, if the caller hangs up before the call is answered, the called user still hears
the ring tone for a period. This problem can be solved by adjusting the send gain on the FXO interface.
Step 5 Run:
impedance { DC value
| AC value }
Applicable Environment
A basic rate access (BRA) interface connects to an ISDN phone. On the AR2200, you can enable
the BRA interface Layer 2 monitoring, remote power supply, automatic deactivation, and alarm
functions, and set the working mode and Layer 1 activation mode on a BRA interface.
Pre-configuration Tasks
Before setting parameters for a BRA interface, complete the following task:
l
Data Preparation
To set parameters for a BRA interface, you need the following data.
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No.
Data
Interface working mode, automatic deactivation delay, and Layer 1 activation mode
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Procedure
Step 1 Run:
system-view
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Applicable Environment
A PRA interface connects to a PBX or PSTN network. On the AR2200, you can enable the
CRC4 check, E1 interface Layer 2 monitoring, and E1 interface pulse code modulation (PCM)
alarm functions, and set the CRC alarm threshold and E1 interface signaling mode on a PRA
interface.
Pre-configuration Tasks
Before setting parameters for a PRA interface, complete the following task:
l
Data Preparation
To set parameters for a PRA interface, you need the following data.
No.
Data
Procedure
Step 1 Run:
system-view
e1t1 e1-voice
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Step 6 Run:
crc4 enable
Pre-configuration Tasks
Before configuring the AR2200 to work in PBX mode, complete the following task:
l
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Procedure
Step 1 Run:
system-view
After the AR2200 is configured to work in PBX mode, restart the AR2200 to make the configuration take
effect.
----End
Context
A SIP AG interface must obtain media and signaling IP addresses from media and signaling IP
address pools respectively. The signaling IP address pool stores IP addresses of PBX interfaces
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and the media IP address pool stores IP addresses of media streams. Media streams and signaling
streams can use the same IP address. Media and signaling IP addresses must be available and
routes are reachable.
Procedure
Step 1 Run:
system-view
Pre-configuration Tasks
Before setting PBX parameters, complete the following task:
l
Data Preparation
To set PBX parameters, you need the following data.
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No.
Data
Country/Region identifier
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No.
Data
Country code
MWI mode
AC amplitude of the ringing current, frequency of the ringing current, and cadence
ratio
10
Applicable Environment
A country/region identifier is configured on a SIP AG so that user terminals connect to the SIP
AG can comply with the local standard.
Procedure
Step 1 Run:
system-view
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Procedure
Step 1 Run:
system-view
The AR does not support the user-defined country code and region code. If the user-defined country code
and region code are used, communication may fail.
To configure the prefix of an international toll call, specify international-prefix international-prefixvalue.
To configure the prefix of a national toll call, specify national-prefix national-prefix-value.
Step 4 Run:
pbx { default-country-code dcc-value | default-area-code dac-value }*
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Procedure
Step 1 Run:
system-view
Context
Hookflash or flash is a button on a telephone that simulates quickly hanging up and then picking
up again (a quick off-hook/on-hook/off-hook cycle). The hookflash can be pressed by a calling
party or a called party:
l
Hookflash pressed by a called party: If the called party user A wants to transfer an incoming
call to user B, user A can press the hookflash and dial the number of user B.
Hookflash pressed by a calling party: User A calls user B. User B answers the call and talks
with user A. User A can press the hookflash and dial the number of user C after hearing a
special dial tone.
If user C is busy, user A can press the hookflash and talk with user B.
If user C does not respond for a long period of time, user A can press the hookflash and
talk with user B.
If the phone of user C rings, user A hangs up and user B hears the ringback tone. User C
picks up the phone and talks with user B.
Whether a called party can be transferred to a toll call is restricted by the outgoing right of
the called party.
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Procedure
Step 1 Run:
system-view
The lower threshold for hookflash pressing must be 50 ms less than the upper threshold for hookflash
pressing.
----End
Context
If there are leave messages, the user device configured with the MWI function makes the
indicator on or plays a tone, indicating that there are leave messages. You can set the MWI mode
according to user habits.
Procedure
Step 1 Run:
system-view
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Context
G.711, also known as Pulse Code Modulation (PCM), is a commonly used waveform codec. G.
711 defines two main compression algorithms, the -law algorithm (used in North America &
Japan) and A-law algorithm (used in Europe and China). A-law encoding takes a 13-bit signed
linear audio sample as input. -law encoding takes a 15-bit signed linear audio sample as input.
Procedure
Step 1 Run:
system-view
Context
Different countries and regions use different ringing standards. You can set the AC amplitude
of the ringing current to adjust the ringing tone volume, voice pitch, cadence ratio, and initial
ringing function on the AR2200 to meet local standards.
Procedure
Step 1 Run:
system-view
Run:
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Run:
user-defined-ring ring-index { first-ring first-ring-period | first-interval
first-interval | second-ring second-ring-period | second-interval secondinterval | third-ring third-ring-period | third-interval third-interval }*
Run:
stop-initial-ring { enable | disable }
Context
When the CLIP service is registered, CLIP parameters in offhook state need to be configured
on the AR2200 so that the AR2200 can work with the phone terminal. Generally, default
parameter settings are used. If CLIP parameters are not set properly, change relevant CLIP
parameters.
Procedure
Step 1 Run:
system-view
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Step 2 Run:
voice
Run:
clip offhook ack-fsk-interval ack-fsk-interval
The interval between the time when the ACK message is received and the time when the
frequency-shift keying (FSK) is transmitted in offhook state is set.
l
Run:
clip offhook dtas-ack-interval dtas-ack-interval
The maximum duration between the time when the dual tone-alerting signal (DT-AS) is
transmitted and the time when the ACK message is received in offhook state is set.
l
Run:
clip offhook dtas-duration dtas-dur-value
The duration of the dual tone-alerting signal (DT-AS) in offhook state is set.
l
Run:
clip offhook dtas-level dtas-level
Run:
clip offhook fsk-level fsk-level
Run:
clip offhook mark-signal-bit mark-signal-bit
The number of bits of the FSK synchronization mask in offhook state is set.
----End
Context
When the CLIP service is registered, CLIP parameters in onhook state need to be configured on
the AR2200 so that the AR2200 can work with the phone terminal. Generally, default parameter
settings are used. If CLIP parameters are not set properly, change relevant CLIP parameters.
Procedure
Step 1 Run:
system-view
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Run:
clip onhook channel-seize-bit channel-seize-bit
Run:
clip onhook dtas-duration dtas-dur-value
Run:
clip onhook dtas-fsk-interval dtas-fsk-interval
The interval between the time when the DT-AS is transmitted and the time when the FSK
is transmitted in onhook state is set.
l
Run:
clip onhook dtas-level dtas-level
Run:
clip onhook fsk-level fsk-level
Run:
clip onhook mark-signal-bit marksignal-bit
The number of bits of the FSK synchronization mask in onhook state is set.
----End
Context
The AR2200 provides uplink bandwidth control. When the system detects that the uplink
bandwidth usage reaches the configured upper threshold, it restricts calls and generates an alarm.
If the uplink bandwidth is insufficient, the system processes calls based on user levels. Common
users may not obtain services.
Procedure
Step 1 Run:
system-view
Run:
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media-bandwidth-control enable
The AR2200 is enabled to restrict calls when the uplink bandwidth is enabled.
l
Run:
media-bandwidth-control maximum max-bandwidth
Run:
media-bandwidth-control reserved-for-emergency reserved-bandwidth
The reserved bandwidth for emergency calls must be smaller than the maximum uplink bandwidth
configured by the media-bandwidth-control maximum command.
----End
Procedure
l
Run the display voice configuration command to check the voice configuration.
Run the display voice sip-reg-count-per-second command to check the number of SIP
Register messages initiated per second.
----End
Applicable Environment
SIP is an IETF-defined signaling protocol widely used for controlling communication sessions
such as voice and video calls over Internet Protocol (IP). SIP, RTP, RTCP, RTSP, and other
protocols constitute a SIP protocol stack.
Pre-configuration Tasks
Before setting SIP protocol stack parameters, complete the following task:
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Data Preparation
To set SIP protocol stack parameters, you need the following data.
No.
Data
Procedure
Step 1 Run:
system-view
Run:
entity-based-sessions-timer enable
Run:
min-se min-seperiod
Run:
session-progress session-progressperiod
Run:
session-rel-progress session-rel-progressperiod
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Run:
t1 t1period
Run:
t2 t2period
Run:
t4 t4period
Run:
td tdperiod
Procedure
Step 1 Run:
system-view
Run:
header-folding enable
Run:
field-header max-forwards max-forwards-value
Run:
field-header organization organization-head
Run:
field-header server server-head
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Run:
field-header user-agent user-agent-head
Procedure
l
----End
Applicable Environment
The digital signal processing (DSP) collects, converts, filters, measures, enhances, compresses,
or identifies signals and coverts the signal from an analog to a digital form.
The DSP module converts analog voice signals into digital signals and stores a certain number
of digital signals into packets for transmission. To improve the voice communication quality,
the DSP needs to further process voice signals.
Pre-configuration Tasks
Before setting DSP parameters, complete the following task:
l
Data Preparation
To set DSP parameters, you need the following data.
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No.
Data
Default DSP channel code type and default interval at which the DSP channel
packetizes RTP packets
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No.
Data
T.30 redundancy parameter value of the T.38 fax, T.4 redundancy parameter value
of the T.38 fax, fax training mode, and maximum fax training rate
Alarm threshold of the dynamic jitter buffer, initial value of the dynamic jitter buffer,
maximum value of the dynamic jitter buffer, maximum value of the static jitter buffer,
minimum value of the dynamic jitter buffer, minimum value of the static jitter buffer,
and initial value of the static jitter buffer of a DSP channel
RTP payload type value, G.726-16k payload type value, G.726-24k payload type
value, G.726-32k payload type value, G.726-40k payload type value, NTE payload
type value, redundancy payload type value, and VBD payload value of a DSP channel
Interval at which a DSP channel sends RTCP packets and threshold for the number
of severe degrade seconds
Data event transmission mode, special process, DTMF transmission mode, echo
cancellation function, input gain, output gain, jitter buffer mode, NLP mode, and DSP
working mode in a DSP template
Context
A user may hear the user's echo in the phone receiver in a conversation. If a proper delay in the
transmitted or received signal is set, the echo can be removed. If the delay exceeds 25 ms, the
voice quality deteriorates and the conversation ends. You can enable echo cancellation on a DSP
channel to remove echoes.
Procedure
Step 1 Run:
system-view
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Context
PLC is a technique that masks the effects of packet loss in VoIP communications. PLC is
effective only when the packet loss ratio is low. During communication, the average packet loss
ratio may be low, but a high burst packet loss ratio results in severe voice quality deterioration.
PLC can insert a static frame in the place where a packet is lost, regenerate a packet received
prior to the lost one, or generate an analog voice packet. If packets are lost during communication
and PLC is not used, the voice communication is interrupted. You can use a proper PLC
algorithm to minimize effects of packet losses.
Procedure
Step 1 Run:
system-view
Context
To save network bandwidth, enable silence compression on a DSP channel. When no voice is
detected, the encoder generates short silence codes, but does not generate voice compression
codes. In addition, the encoder notifies the receiver of silence start until the voice is restored.
The silence compression function reduces the number of sent voice packets.
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Procedure
Step 1 Run:
system-view
----End
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Procedure
Step 1 Run:
system-view
Run:
fax redundancy-t4 redundancy-t4value
Run:
fax redundancy-t30 redundancy-t30value
Run:
fax training-mode { e2e | local }
Run:
fax training-rate { v17 | v29 | v27 }
Run:
fax v8negotiate enable
V8 negotiation is enabled.
----End
Context
Delay variations in voice packet arrival time can occur because of network congestion or route
changes. To reduce sound distortion caused by the delay jitter and packet loss, a jitter buffer is
used. You can set proper jitter buffer parameters to minimize delay variations so that packets
can be processed in a timely manner and smooth voice communication can be provided as much
as possible.
Procedure
Step 1 Run:
system-view
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Run:
jitter-buffer adapt-jb-threshold adapt-jb-threshold
Run:
jitter-buffer init-adapt-jb init-adapt-jb-value
Run:
jitter-buffer init-fixed-jb normal-fixed-jb-value
Run:
jitter-buffer max-adapt-jb max-adapt-jb-value
Run:
jitter-buffer min-adapt-jb min-adapt-jb-value
Run:
jitter-buffer max-fixed-jb max-fixed-jb-value
Run:
jitter-buffer min-fixed-jb min-fixed-jb-value
Procedure
Step 1 Run:
system-view
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Run:
payload-type clear-mode clearmode-value
Run:
payload-type g726-16k g726-16k-value
Run:
payload-type g726-24k g726-24k-value
Run:
payload-type g726-32k g726-32k-value
Run:
payload-type g726-40k g726-40k-value
Run:
payload-type nte nte-value
Run:
payload-type redundancy redundancy-value
Run:
payload-type vbd vbd-value
Context
RTCP monitors the quality of service and conveys information about participants in an on-going
session. RTCP periodically sends packets to all the participants in the session to monitor the
quality of service and obtain identity information about the participants.
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Procedure
Step 1 Run:
system-view
Run:
rtcp rtcp-interval rtcp-interval
Run:
rtcp rtcpxr enable
The RTP Control Protocol Extended Reports (RTCP XR) function is enabled.
l
Run:
rtcp sev-degradethreshold sev-degradethresholdval
Run:
rtcp vqm enable
Context
DSP resources are limited and users have different requirements for DSP resources. To control
and allocate DSP resources properly, set the DSP resource control mode and the resource
threshold in hierarchical control mode.
Procedure
Step 1 Run:
system-view
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Run:
resource-threshold mode { priority | normal }
Run:
resource-threshold { threshold1 threshold1 | threshold2 threshold2 |
threshold3 threshold3 }
Context
To customize DSP parameters for data services, configure a DSP template. After a DSP template
is configured, specify the template for users according to the port and phone number. The DSP
template improves the call connection rate. After a template is specified successfully, parameters
in the DSP template take effect immediately.
Procedure
Step 1 Run:
system-view
Run:
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data-event data-event-value
Run:
dsp-special-flow dsp-special-flow-value
Run:
dtmf dtmf-value
Run:
echo-cancel { enable | disable }
Run:
input-gain input-gain-value
Run:
output-gain output-gain-value
Run:
jitter-buffer { dynamic | static }
Run:
nlp nlp-value
Run:
work-mode work-mode-value
Run:
vbd redundancy
Context
You can enable a digital signal processor (DSP) channel to work in loopback mode, and set the
loopback mode (PCM-side loopback test and IP-side loopback test). When the DSP channel
between the calling party and called party cannot transmit signals or can transmit signals only
in one direction, run the loop-back command to locate the fault. If the calling party hears the
echo in a PCM-side loopback test, the speech channel between the calling phone and the calling
DSP channel is functioning properly. If the called party hears the echo in an IP-side loopback
test, the speech channel between the called phone and the calling DSP channel is functioning
properly.
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To control resources of DSP channels, prohibit the DSP channels. The prohibited DSP channels
cannot participate in resource allocation.
Procedure
Step 1 Run:
system-view
Run:
loop-back loopback-type channel
Run:
prohibit channel [ count ]
Procedure
l
Run the display voice dsp-attribute command to check the DSP configuration.
Run the display voice dsp-template command to check the DSP template configuration.
----End
Applicable Environment
When multiple enterprises access a PBX, the PBX can be divided into multiple virtual PBXs so
that the enterprises can use one PBX. Configuring enterprises on the PBX facilitate user
management.
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Pre-configuration Tasks
Before configuring an enterprise, complete the following task:
l
Data Preparation
To configure an enterprise, you need the following data.
No.
Data
Enterprise name
DN set
Procedure
Step 1 Run:
system-view
By default, a maximum of 16 Centrex groups can be configured in the enterprise view for
different devices
l AR2220: 16
l AR2240: 32
Step 6 Run:
dn-set dn-set-name [ description description ]
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A DN set is configured.
By default, the default DN set defaultdialplan is generated after an enterprise is created.
Step 7 (Optional) Run:
description enterprise-name
Usage Scenario
A call prefix, an important attribute of the call service, defines a call number rule and describes
the call number distribution and routing plans in an exchange office.
Pre-configuration Tasks
Before configuring a call prefix, complete the following task:
l
Configuring an enterprise
Data Preparation
To configure a call prefix, you need the following data.
No.
Data
Call prefix profile name, enterprise and dial number (DN) set to be bound to the call
prefix, call type, call attribute, maximum length and minimum length of a number
that can be parsed, (optional) home area attribute, and (optional) ringing delay
Procedure
Step 1 Run:
system-view
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The call type and call attribute are configured for the call prefix.
By default, no call type or call attribute is configured for a call prefix.
Step 7 Run:
digit-length maximum-length-value minimum-length-value
The maximum length of a number that can be parsed must be greater than or equal to the minimum length
of a number that can be parsed.
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2 PBX Configuration
Usage Scenario
When a user is needed for the voice service, you need to create a PBX user on the PBX.
Pre-configuration Tasks
Before configuring a PBX user, complete the following tasks:
l
Configuring an enterprise
Data Preparation
To configure an enterprise, you need the following data.
No.
Data
User name
Procedure
Step 1 Run:
system-view
Run the port slotid/subcard/portid command to configure a port for the PBX user.
Run the sipue eid-value command to configure a device ID for the SIP UE user.
Step 5 Run:
telno [ country-code country-code-value ] [ area-code area-code-value ] telno-value
The country code, area code, and telephone number of the PBX user are configured.
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By default, no country code, area code, or telephone number is configured for a PBX user.
Step 6 Run:
dn-set dn-set-name
Usage Scenario
If SIP users need to communicate with each other through the AR2200, configure a SIP server.
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Pre-configuration Tasks
Before configuring a SIP server, complete the following tasks:
l
Configuring an enterprise
Data Preparation
To configure a SIP server, you need the following data.
No.
Data
Procedure
Step 1 Run:
system-view
A signaling IP address and signaling port number are configured for the SIP server.
By default, no signaling IP address or signaling port number is configured.
Step 5 Run:
signalling-domain signaling-domain-value
The signaling domain name is configured for the SIP server using a dynamic signaling IP address.
NOTE
When a SIP server uses a dynamic signaling IP address, configure a signaling domain name for the SIP
server.
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Step 6 Run:
ddns-client ddns-client-name
A dynamic domain name system (DDNS) name is configured for a SIP server using a dynamic
signaling IP address.
NOTE
When a SIP server uses a dynamic signaling IP address, configure a DDNS client name for the SIP server so
that the DDNS server can update the mapping between the signaling domain name and IP address.
Step 7 Run:
media-ip { ip-address | addr-name addr-name-value }
CAUTION
Exercise caution when you run this command because resetting a SIP server interrupts running
services.
----End
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2 PBX Configuration
Applicable Environment
As shown in Figure 2-5, the SIP server (router A) at the headquarters uses a public IP address,
and traveling SIPUEs and some SIPUEs in the branch use private IP addresses. To provide voice
services through the PBX at the headquarters for SIPUEs, the SBC proxy function must be
configured on router A.
Figure 2-5 SBC proxy configuration networking diagram (public IP address for the SIP server
and private IP addresses for SIPUEs)
Branch
SIP UE
Enterprise A
Router C
SIP trunk
Router B
ISDN
IP network
SIP UE
(Travelling staff)
SIP trunk
Router D
Router A
Headquarters
User A
Enterprise A
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Pre-configuration Tasks
Before configuring the SBC proxy function, complete the following tasks:
l
Configuring IP addresses to enable SIPUEs and SIP servers to communicate with each
other
Data Preparation
To configure the SBC proxy function, you need the following data.
No.
Data
Media trunk type, and signaling and media proxy policies for the SIP server
Procedure
Step 1 Run:
system-view
If all SIPUEs are located on the private network, set the value to Enable. If only some of the SIPUEs are located
on the private network, set the value to Auto.
Step 6 Run:
sbc media-proxy value
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NOTE
If all SIPUEs are located on the private network, set the value to Enable. If only some of the SIPUEs are located
on the private network, set the value to Auto.
----End
Usage Scenario
As shown in Figure 2-6, the SIP server (router B) at the headquarters uses a private IP address,
and SIP UEs in the branch or on business trips use private IP addresses. The SBC proxy function
must be configured for the SIP server to provide voice services using the PBX at the headquarters
for SIP UEs.
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Figure 2-6 SBC proxy configuration networking diagram (private IP address for the SIP server
and public IP addresses for SIP UEs)
Headquarters
SIP UE
Enterpeise A
Router B
SIP trunk
ISDN
Router A
IP network
SIP UE
(Travelling staff)
Pre-configuration Tasks
Before configuring the SBC proxy function, complete the following tasks:
l
Configuring IP addresses to enable SIP UEs and SIP servers to communicate with each
other
Data Preparation
To configure the SBC proxy function, you need the following data.
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No.
Data
Media trunk type, and signaling and media proxy policies for the SIP server
Procedure
Step 1 Run:
system-view
If all SIP UEs are located on the private network, set the value to Enable. If only some of the SIP UEs are located
on the private network, set the value to Auto.
Step 6 Run:
sbc media-proxy value
If all SIP UEs are located on the private network, set the value to Enable. If only some of the SIP UEs are located
on the private network, set the value to Auto.
Step 7 Run:
sbc mapped-signalling-address ip ip-address port port-value
A public IP address and port number are configured for signaling mapping of the SIP server.
Step 8 Run:
sbc mapped-media-ip ip-address
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2 PBX Configuration
Usage Scenario
As shown in Figure 2-7, the SIP server (router A) at the headquarters uses a private IP address,
and SIP UEs in the branch or on business trips use private IP addresses. To provide voice services
through the PBX at the headquarters for SIP UEs, the SBC proxy function must be configured
on router A.
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Figure 2-7 SBC proxy configuration networking diagram (private IP address for the SIP server
and private IP addresses for SIP UEs)
Branch
SIP UE
Enterpeise A
Router C
SIP trunk
Router B
IP network
Headquarters
Router A
SIP trunk
Enterpeise A
Pre-configuration Tasks
Before configuring the SBC proxy function, complete the following tasks:
l
Configuring IP addresses to enable SIP UEs and SIP servers to communicate with each
other
Data Preparation
To configure the SBC proxy function, you need the following data.
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No.
Data
Media proxy port number, media trunk type, and signaling and media proxy policies
for the SIP server
Procedure
Step 1 Run:
system-view
If all SIP UEs are located on the private network, set the value to Enable. If only some of the SIP UEs are
located on the private network, set the value to Auto.
l Run:
sbc media-proxy value
If all SIP UEs are located on the private network, set the value to Enable. If only some of the SIP UEs are
located on the private network, set the value to Auto.
l Run:
sbc mapped-signalling-address ip ip-address port port-value
A public IP address and port number are configured for signaling mapping of the SIP server.
l Run:
sbc mapped-media-ip port-value
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2 PBX Configuration
l Run:
sbc mapped-media-proxy-port-start port-value
The media proxy port number mapping the SIP server is configured.
l Run:
return
If all SIP UEs are located on the private network, set the value to Enable. If only some of the SIP UEs are
located on the private network, set the value to Auto.
l Run:
sbc mapped-signalling-address ip ip-address port port-value
A public IP address and port number are configured for signaling mapping of the SIP server.
l Run:
sbc mapped-media-ip ip-address
The public media IP address mapping the SIP trunk group is configured.
l Run:
sbc mapped-media-port-start port-value
The media start port number mapping the SIP trunk group is configured.
l Run:
sbc mapped-media-proxy-port-start port-value
The media proxy start port number mapping the SIP trunk group is configured.
----End
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Applicable Environment
To record and display CDRs of users, specify a CDR server on the AR2200.
Pre-configuration Tasks
Before configuring a CDR server, complete the following tasks:
l
Data Preparation
To configure a CDR server, you need the following data.
No.
Data
Procedure
Step 1 Run:
system-view
The IP address and port number of the CDR server are specified.
By default, no IP address or port number is configured for a CDR server.
Step 5 Run:
username username password { cipher cipher-password | simple simple-password}
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2 PBX Configuration
Usage Scenario
As shown in Figure 2-8, the AR2200 functions as a PBX and needs to communicate with other
PBXs or external networks. The AR2200 supports the following connection modes:
l
Trunks for inter-office calls with the same attributes are added to a trunk group. A trunk group
is associated with a call route to establish call sessions between exchange offices.
Figure 2-8 Trunk application
PSTN
IMS
SIP/H323
AT0/PRA
AR
SIP/PRA/E1R2/H323
PBX
Pre-configuration Tasks
Before configuring a trunk group, complete the following tasks:
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Data Preparation
To configure a trunk group, you need the following data.
No.
Data
Common parameters of a trunk group: trunk group name, signaling type, DN set and
enterprise to be bound to the trunk group, and optional parameters including the callin right, call-out right, country code, area code, default number displayed, trunk
circuit selection mode, and trunk group description
Parameters of a SIP trunk group: SIP registration mode, home domain of the peer
SIP trunk, signaling IP address or dynamic signaling IP address name of the SIP trunk
group, media IP address or dynamic media IP address name of the SIP server,
signaling port of the local end, IP address and port of the remote end, registrar
Uniform Resource Identifier (URI), and optional parameters including the
authentication password, trunk group identifier, maximum number of concurrent
calls, transmission mode, numeral software parameter and its index, Dual-Tone
Multi-frequency (DTMF) parameters, fax/modem codec negotiation mode,
packetization interval, voice band data (VBD) attribute type, VBD codec mode, VBD
payload type, modem transmission mode, RFC 2198 negotiation start mode, RFC
2198 redundancy transmission start mode, RFC 2833 negotiation start mode, and
RFC 2833 fax and modem transmission
Parameter of a PRA trunk group: protocol used
Parameters of a SIP AT0 trunk: ID and password for registration, trunk name, called
number of incoming calls, and calling number of outgoing calls
Parameters of an AT0 trunk: trunk name, location of the Foreign Exchange Office
(FXO) interface connected to the trunk, signal transmission type for the calling line
identification presentation (CLIP) service, dial delay, call prefix, dial delay after the
call prefix is inserted, called number of incoming calls, dialing mode, and trunk status
Parameters of a PRA trunk: trunk name, and location of the physical interface
connected to the trunk
Parameters of a PRA trunk: trunk name and member interfaces in the PRA trunk
Parameters of an E1R2 trunk: R2 signaling type, sending parameter of R2 line
signaling and register signaling, receiving parameter of R2 line signaling and register
signaling, line signaling and register signaling of R2 profile, register's receiving
address of R2 profile, signaling profile name of E1R2, and multiple-language
adaptive signaling of R2 profile
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Context
When the signaling type of a trunk group is dss1-net, dss1-user, qsig-net, or qsig-user, the trunk
group is a PRA trunk group.
When the signaling type of a trunk group is FXO, the trunk group is an AT0 trunk group.
When the signaling type of a trunk group is SIP, the trunk group is a SIP trunk group.
When the signaling type of a trunk group is e1-r2, the trunk group is an E1R2 trunk group.
When the signaling type of a trunk group is H323, the trunk group is an H323 trunk group.
In PBX mode:
l
When the AR2200 functions as a network-side device and uses DSS1 signaling to connect
to the remote network through a PRA trunk, set the signaling type to dss1-net.
When the AR2200 functions as a user-side device and uses DSS1 signaling to connect to
the remote network through a PRA trunk, set the signaling type to dss1-user.
When the AR2200 functions as a network-side device and uses QSIG signaling to connect
to the remote network through a PRA trunk, set the signaling type to qsig-net.
When the AR2200 functions as a user-side device and uses QSIG signaling to connect to
the remote network through a PRA trunk, set the signaling type to qsig-user.
When the AR2200 connects to the remote network through an AT0 trunk, set the signaling
type to FXO.
When the AR2200 connects to the remote network through a SIP trunk, set the signaling
type to SIP.
If the AR2200 uses R2 signaling to connect to a network through the PRA trunk and is used
as a user-side device, the signaling type is set to e1-r2.
Procedure
Step 1 Run:
system-view
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2 PBX Configuration
system-view
The incoming and outgoing call rights are configured for the trunk group.
l Run:
default-caller-telno country-code-value area-code-value value
The country code, area code, and default number displayed are configured for the trunk group.
l Run:
description desc-value
Procedure
Step 1 Run:
system-view
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Procedure
Step 1 Run:
system-view
The home domain to which the SIP trunk of the peer device belongs is configured.
l Run:
signalling-address { ip ip-address | addr-name signal-addr-name-value } port
port-value
A signaling IP address and port number are configured for the SIP trunk group.
l Run:
media-ip {ip-address | addr-name addr-name-value}
The media IP address or dynamic media IP address name is configured for the SIP server.
l Run:
peer-address static primary-ip-value primary-port-value [ secondary
ip-value secondary-port-value ]
secondary-
The remote IP address and port are configured for the SIP trunk group.
l Run:
register-uri register-uri-value
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The trunk authentication password and trunk group identifier need to be configured only when the SIP
trunk registration mode is trunk group registration.
The numeral software parameter and numeral software parameter index are set.
l Run:
dtmf-transmission-mode { thoroughly | erase }
The mode in which RFC 2198 redundancy transmission negotiation is started is set.
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l Run:
redundancy-start-mode { ordinary2198 | smart2198 }
The PBX is enabled to transmit VBD using RFC 2198 redundancy transmission.
l Run:
nte-flashhook enable
Peer mode
As shown in Figure 2-9, RouterA and RouterB are enterprise routers that can communicate.
Voice services between RouterA and RouterB are transmitted through the H323 trunk. The
H323 trunk uses the peer mode, so it does not need to be registered with the gatekeeper.
Figure 2-9 Networking in peer mode
N e tw
IP o rk
U s e rA
R o u te rA
GW
R o u te rB
GW
U se rB
G W s a re p e e rs
Registration mode
As shown in Figure 2-10, the gatekeeper is the carrier device and the gateway is the
enterprise device. The gateway can register with the gatekeeper to transmit voice services.
The gatekeeper needs to identify the registered gateway. After the gateway is registered on
the gatekeeper, the gatekeeper completes voice services initiated by the gateway. The
AR2200 does not function as the gateway, that is, the AR2200 does not send registration
packets.
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IP o rk
N e tw
R o u te rB
GW
U se rA
G a te w a y re g iste rs w ith th e
g a te ke e p e r
GW
GK
R o u te rA
GK
Procedure
Step 1 Set H323 attributes.
l (Mandatory) Set H323 attribute.
Run:
system-view
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Action
Command
Remarks
Configure the
Q931
listening port.
q931-listenport q931listenport-value
Configure the
AR2200
whether to
enable fast
start.
NOTE
After the fast start function is
enabled, sessions can be set up
quickly.
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2 PBX Configuration
Action
Command
Remarks
Configure the
AR2200
whether to
support the
H245 control
signaling
channel.
Set the
operation
mode for the
H323 stack.
workmode workmode-value
Configure the
UDP port
number of
RAS signals.
rasport rasport-value
Configure the
H323 stack
system name.
h323-systemname h323systemnamevalue
Configure the
H323 stack
product name.
productname productname-value
Configure the
H323 stack
version.
h323version h323version-value
Configure the
H245 version.
h245version h245version-value
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Action
Command
Remarks
Configure the
display
header field in
Q931
messages.
displayname displayname-value
tcp-keepalive-timerlen tcp-keepalivetimerLen-value
Configure the
AR2200
whether to
support
automatic
setup of the
H245
channel.
Configure the
H323 stack
whether to
receive the
H245 address
in Setup
messages.
Set the
timeout
interval for
TCP
connections.
tcp-connection-timeout tcp-connectiontimeout-value
Set the
timeout
interval for
Setup
messages.
setup-wait-response-timeout setupwait-response-timeout-value
Set the
timeout
interval for
Alerting and
Connect
messages.
callproc-wait-response-timeout
callproc-wait-response-timeout-value
Set the
timeout
interval for
Connect
messages.
recvalert-wait-response-timeout
recvalert-wait-response-timeout-value
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2 PBX Configuration
Action
Command
Remarks
Set the
channel
establishment
timeout.
channels-timeout channels-timeoutvalue
Set the
timeout
interval of the
roundtrip
delay process.
roundtrip-delay-timeout roundtripdelay-timeout-value
Set the
timeout
interval for
mode
switching.
requestmode-timeout requestmodetimeout-value
Set the
timeout
interval of the
media loop
process.
medialoop-timeout medialoop-timeoutvalue
Set the
timeout
interval for
the peer end to
send
EndSessionC
ommand
messages.
endsession-timeout endsession-timeoutvalue
l Run:
reset
After setting H323 system parameters, run the reset command to reset the H323 system to make the
parameters take effect.
l Run:
quit
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The media IP address or dynamic media IP address name is configured for the H323 trunk
group.
l Run:
gwid gwid-value
The gateway ID of the peer device connected to the H323 trunk is configured.
NOTE
When the AR2200 functions as the gatekeeper, you must configure the gateway ID of the peer device.
l Run:
peer-address static primary-ip-value primary-port-value
The remote IP address and port number are configured for the remote H323 trunk of the H323
trunk group.
NOTE
When the AR2200 functions as the gateway, you must configure the remote IP address and port number.
l When there are trunk groups of various H323 types, perform the following operations.
Run:
techprefix techprefix-value
Procedure
Step 1 Run:
system-view
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2 PBX Configuration
voice
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2 PBX Configuration
quit
The E1R2 signaling profile used by the E1R2 trunk group is configured.
Step 9 Run:
r2-receive-earlymedia r2-receive-earlymedia
Whether to enable the early media for E1R2 trunk group is configured.
Step 10 Run:
r2-play-ringback r2-play-ringback
Whether to enable the ringback tone (RBT) for E1R2 trunk group is configured.
----End
Procedure
Step 1 Run:
system-view
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2 PBX Configuration
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2 PBX Configuration
Procedure
Step 1 Run:
system-view
After the E1R2 trunk group is complete, to ensure that call services passing through the E1R2 trunk group
are transmitted correctly, local loopback cannot be configured on the VE1 interface. If local loopback has
been configured on the VE1 interface, cancel the local loopback configuration immediately. One minute
after the local loopback configuration is canceled, call services passing through the E1R2 trunk group are
restored.
----End
Applicable Environment
To make modified SIP trunk group parameters take effect or restart a new SIP trunk group, reset
the SIP trunk group.
Pre-configuration Tasks
Before resetting a SIP trunk group, complete the following task:
l
Procedure
Step 1 Run:
system-view
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2 PBX Configuration
Step 3 Run:
trunk-group name
CAUTION
Exercise caution when you run this command because resetting a SIP trunk group affects running
services.
----End
Procedure
l
Run the display voice trunk-group command to check the configuration of a trunk group.
----End
Applicable Environment
As shown in Figure 2-11, the headquarters and branch of enterprise A are connected through a
SIP trunk. The headquarters is connected to PSTN networks of carrier A and carrier B. To reduce
call fees, configure call routes on the AR2200, which meet the following requirements:
l
Calls between users in the headquarters and branch are connected through the SIP trunk.
When users in the enterprise call external users, the calls are connected to the called users
through the network of carrier A or carrier B, depending on the charge rates used by the
carriers in different time ranges.
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Branch
User B
PSTN
Carrier A
AR
Enterprise A
SIP
PSTN
Carrier B
IP
SIP
PRA
AT0
AR
Headquarters
PBX
Enterprise A
User A
Pre-configuration Tasks
Before configuring a call route, complete the following tasks:
l
Data Preparation
To configure a call route, you need the following data.
Issue 01 (2012-04-20)
No.
Data
Enterprise, Centrex group, and DN set to which the calling number is bound, and
calling condition
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No.
Data
2 PBX Configuration
Procedure
Step 1 Run:
system-view
Applicable Environment
After the abbreviated dialing service is configured on the AR2200, users can dial the 2-digit
abbreviated code instead of the original called number, which is convenient for users to operate
and manage the phone numbers.
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Pre-configuration Tasks
Before configuring the abbreviated dialing service, complete the following tasks:
l
Data Preparation
To configure the abbreviated dialing service, you need the following data.
No.
Data
Abbreviated code
Procedure
Step 1 Run:
system-view
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2 PBX Configuration
Applicable Environment
To restrict outgoing calls such as international toll calls, configure the call-out restriction service.
Pre-configuration Tasks
Before configuring the call-out restriction service, complete the following tasks:
l
Data Preparation
To configure the call-out restriction service, you need the following data.
No.
Data
User name
Procedure
Step 1 Run:
system-view
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2 PBX Configuration
Usage Scenario
To ensure that users can answer incoming calls, configure the following types of call forwarding
service:
l
Call Forwarding Busy (CFB): Incoming calls of a CFB user are forwarded to a preset
number when the user line is busy.
Call Forwarding No Reply (CFNR): Incoming calls of a CFNR user are forwarded to a
preset number when the user does not answer the call.
Call Forwarding Offline (CFO): Incoming calls of a CFO user are forwarded to a preset
number when the user is offline.
Call Forwarding Unconditional (CFU): All incoming calls of a CFU user are forwarded to
a preset number.
Pre-configuration Tasks
Before configuring the call forwarding service, complete the following tasks:
l
Data Preparation
To configure the call forwarding service, you need the following data.
No.
Data
Forwarded-to number
Procedure
Step 1 Run:
system-view
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2 PBX Configuration
Run the display voice [ pbxuser name ] service cfb command to check the CFB service.
Run the display voice [ pbxuser name ] service cfb command to check the CFNR service.
Run the display voice [ pbxuser name ] service cfo command to check the CFO service.
Run the display voice [ pbxuser name ] service cfu command to check the CFU service.
Pre-configuration Tasks
Before configuring the number barring service, complete the following tasks:
l
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Data Preparation
To configure the number barring service, you need the following data.
No.
Data
Restricted number
Procedure
Step 1 Run:
system-view
Applicable Environment
The DND service allows a user to reject all incoming calls during a certain period.
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Pre-configuration Tasks
Before configuring the DND service, complete the following tasks:
l
Data Preparation
To configure the DND function, you need the following data.
No.
Data
DND tone
Procedure
Step 1 Run:
system-view
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2 PBX Configuration
Applicable Environment
To prevent anonymous calls, configure the RAC service.
Pre-configuration Tasks
Before configuring the RAC service, complete the following tasks:
l
Data Preparation
To configure the RAC service, you need the following data.
No.
Data
RAC tone
Procedure
Step 1 Run:
system-view
151
2 PBX Configuration
Applicable Environment
The remote office service allows a user to access from any terminal and share original services
such as short number dialing and call transfer. Because the call initiator still dials the original
number in the remote office service, the user privacy is well protected.
Pre-configuration Tasks
Before configuring the remote office service, complete the following tasks:
l
Data Preparation
To configure the remote office service, you need the following data.
No.
Data
Procedure
Step 1 Run:
system-view
152
2 PBX Configuration
Step 4 Run:
service-right remote-office enable
Applicable Environment
You can configure the secretary service to filter incoming calls and prevent interruption.
Pre-configuration Tasks
Before configuring the secretary service, complete the following tasks:
l
Data Preparation
To configure the secretary service, you need the following data.
No.
Data
Procedure
Step 1 Run:
system-view
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Applicable Environment
When the wake-up time is due, the system sends the wake-up tone to the user.
Pre-configuration Tasks
Before configuring the wake-up service, complete the following tasks:
l
Data Preparation
To configure the wake-up service, you need the following data.
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No.
Data
User name
Wake-up time
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2 PBX Configuration
Procedure
Step 1 Run:
system-view
Usage Scenario
After a user registers the RBT service, the user can set different RBTs for a calling party or a
group of calling parties in different periods.
Pre-configuration Tasks
Before configuring the RBT service, complete the following tasks:
l
Data Preparation
To configure the RBT service, you need the following data.
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No.
Data
2 PBX Configuration
Context
If the RBT played for the caller through the FXO port is too loud, the FXO port may fail to detect
the busy tone from the caller; therefore, if the caller hangs up before the call is answered, the
called user still hears the ring tone for a period. To solve this problem, run the gain send send
command to adjust the send gain on an FXO interface.
Procedure
Step 1 Run:
system-view
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2 PBX Configuration
Usage Scenario
If dialing some numbers is prevented, configure the SCR service.
Pre-configuration Tasks
Before configuring the SCR service, complete the following tasks:
l
Data Preparation
To configure the SCR service, you need the following data.
No.
Data
Rejected number
Procedure
Step 1 Run:
system-view
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2 PBX Configuration
Usage Scenario
To receive only some incoming calls, configure the SCA service.
Pre-configuration Tasks
Before configuring the SCA service, complete the following tasks:
l
Data Preparation
To configure the SCA service, you need the following data.
No.
Data
Procedure
Step 1 Run:
system-view
158
2 PBX Configuration
Step 3 Run:
pbxuser name
Usage Scenario
A call may fail to be connected for some reasons, for example, the called user is busy, the called
user does not answer the call, the called user is offline, the called number does not exist, the
calling user does not have right to make this call, or the called user is unreachable. If the call
interception service is not configured, the calling user only hears the busy tone but does not
know the reason. If the user redials the called number multiple times but the call still fails to be
connected, the user experience is degraded and network resources are wasted. The call
interception service allows the system to play a user-friendly voice prompt when a call failed to
be connected. The called user knows the cause of the call failure and determines whether to
redial the called number according to the voice prompt. This improves the user experience,
reduces invalid dial attempts, and saves resources on the AR2200.
Pre-configuration Tasks
Before configuring the call interception service, complete the following tasks:
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2 PBX Configuration
Data Preparation
To configure the call interception service, you need the following data.
No.
Data
Call interception plan name, enterprise, DN set, and call prefix to be bound to the
call interception plan, and call status.
Procedure
Step 1 Run:
system-view
A call interception service plan is created and the call interception service view displayed.
By default, no call interception service plan is configured.
NOTE
Before creating a call interception plan, run the service-right call-intercept enable command in the PBX
user view to enable the call interception service right.
Step 4 Set mandatory parameters for the call interception service plan.
l Run:
enterprise enterprise-name [ dn-set dn-set-name | centrex centrex-name ]
An enterprise, a DN set, and a Centrex group are bound to the call interception service plan.
l Run:
callprefix callprefix-name
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2 PBX Configuration
Applicable Environment
User A and user B belong to the same Centrex group, and user C is out of the Centrex group. If
the distinctive ringing service is not configured, the same ring tone is played to user B no matter
whether user A or user C calls user B. User B cannot know whether the call is from a Centrex
user or an external user according to the ring tone. After the distinctive ringing service is
configured, users in the Centrex group can identify calls from users in the same Centrex group
and outside the Centrex group.
Pre-configuration Tasks
Before configuring the distinctive ringing service, complete the following tasks:
l
Creating an enterprise
Data Preparation
To the distinctive ringing service, you need the following data.
No.
Data
Distinctive ringing service plan name, and enterprise and Centrex group bound to the
plan
(Optional) Ring IDs for intra-group calls, local calls, national toll calls, and
international toll calls
Procedure
Step 1 Run:
system-view
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A distinctive ringing service plan is created and the distinctive ringing service view is displayed.
By default, no distinctive ringing service plan is configured.
Step 4 Set mandatory parameters for the distinctive ringing service plan.
l Run:
enterprise enterprise-name [ centrex centrex-name ]
Usage Scenario
If the RBT service is not configured for an enterprise, a user only hears the common ring back
tone when calling a user in the enterprise. With the enterprise RBT service, the enterprise can
play an advertisement or enterprise information to calling users. The enterprise can also
configure different RBTs for customers. The enterprise RBT service improves user experience
and helps the enterprise raise brand image.
Pre-configuration Tasks
Before configuring the enterprise RBT service, complete the following tasks:
l
Creating an enterprise
Data Preparation
To configure the enterprise RBT service, you need the following data.
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No.
Data
Name of the RBT service plan and enterprise bound to the plan
2 PBX Configuration
Context
If the RBT played for the caller through the FXO port is too loud, the FXO port may fail to detect
the busy tone from the caller; therefore, if the caller hangs up before the call is answered, the
called user still hears the ring tone for a period. This problem can be solved by adjusting the
send gain on the FXO interface.
Procedure
Step 1 Run:
system-view
Before configuring the enterprise RBT service, run the service-right crbt enable command in the PBX user
view to enable the enterprise RBT service.
----End
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2 PBX Configuration
Applicable Environment
The interactive voice response (IVR) service provides the automated attendant function and
allows enterprises to make their own IVR menus and voice prompts.
Pre-configuration Tasks
Before configuring an IVR service, complete the following task:
NOTE
If the codec mode is set to G.723, access terminals cannot use the IVR service.
Data Preparation
To configure the IVR service, you need the following data.
Issue 01 (2012-04-20)
No.
Data
IVR menu name, type, enterprise, prompt tone, and optional parameters, including
the menu tone, prompt tone played when there is no input on the IVR menu within
the specified period, prompt tone played when the input information does not match
the preconfigured information of the IVR menu, prompt tone played when the number
of incorrect inputs on the IVR menu reaches the maximum value, tone interruption
flag, whether to replay the prompt tone when there is no input on the IVR menu within
the specified period, whether to replay the prompt tone when the input information
does not match the preconfigured information of the IVR menu, wait duration, and
maximum number of times the input information does not match the preconfigured
information of the IVR menu
IVR action menu name, current action status of the menu, operation code, IVR action
code, and optional parameters, including the prompt tone, sub-menu, forward-to
number, DN set of the forward-to number, Centrex group, and enterprise of the
forward-to number
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2 PBX Configuration
No.
Data
Enterprise, DN set, access number, validity period, validity cycle mode, calling
number, switchboard number, and optional parameters, including the prompt tone,
destination enterprise name used when the IVR group connects a call to an extension
phone number, maximum queue duration and maximum number of calls in a queue,
(optional) ringing duration and ringing mode, flag indicating whether the IVR group
can directly connect to the switchboard, and menu name of the IVR group
Enterprise, DN set, and optional parameters, including the maximum queue duration
and maximum number of calls in a queue and ringing duration and ringing mode of
the NAVI group
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Procedure
Step 1 Run:
system-view
| ivr-file name }
When the IVR menu is a root menu, this step can be skipped.
The prompt tone played when there is no input on the IVR menu within the specified period
is configured.
l Run:
no-match-tone { toneid | ivr-file name }
The prompt tone played when the input information does not match the preconfigured
information of the IVR menu.
l Run:
max-error-tone { toneid | ivr-file name }
The prompt tone played when the number of incorrect inputs on the IVR menu reaches the
maximum value is configured.
l Run:
menu-bargein menubargein-value
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2 PBX Configuration
The device determines whether to replay the prompt tone when there is no input on the IVR
menu within the specified period.
l Run:
no-match-reprompt { enable | disable }
The device determines whether to replay the prompt tone when the input information does
not match the preconfigured information of the IVR menu.
l Run:
menu-wait-timer value
The maximum number of times the input information does not match the preconfigured
information of the IVR menu is set.
----End
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2 PBX Configuration
The IVR menu status displayed after the IVR action is taken is configured.
l Run:
destination enterprise destination-enterprise-name [ dn-set dn-set-name |
centrex centrex-name ]*
The destination enterprise name, destination DN set, and destination Centrex group are
configured for the forward-to number of the IVR action.
----End
The calling party, validity period, and time repeat mode are configured for the IVR group.
Step 7 Run:
console-telno value
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2 PBX Configuration
The destination enterprise name used when the IVR group connects a call to an extension
phone number is configured.
l Run:
queue enable [ maximum-queue
value ]
The maximum queue duration and maximum number of calls in a queue are set.
l Run:
ring { mode ring-mode-value | time ring-time-value | select select-mode-value }
| enable }
The flag indicating whether the IVR group can connect to the switchboard is set.
l Run:
navigate-menu menu-name
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An operation code input when the IVR navigation group takes the menu action is configured.
Step 7 Set optional parameters for the IVR navigation group.
l Run:
queue enable [ maximum-queue
value ]
The maximum queue duration and maximum number of calls in a queue are set.
l Run:
ring { mode ring-mode-value | time ring-time-value
value }
| select
select-mode-
Procedure
l
Run the display voice ivr-action [ ivr-action-name ] command to view information about
one IVR action or all IVR actions.
Run the display voice ivr-file [ ivr-file-name ] command to view information about one
IVR file or all IVR files.
Run the display voice ivr-menu [ ivr-menu-name ] command to view information about
one IVR menu or all IVR menus.
----End
Usage Scenario
To hide calling numbers or display the same calling number for all outgoing calls, configure the
number change service.
Pre-configuration Tasks
Before configuring the number change service, complete the following tasks:
l
Data Preparation
To configure the number change service, you need the following data.
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2 PBX Configuration
No.
Data
Enterprise and DN set bound to the number change plan, calling number that needs
to be changed, and number change rule
Procedure
Step 1 Run:
system-view
Usage Scenario
You can configure pre-routing number change plans to define various dialing modes and change
the calling number displayed on the called party's phone. For example, a POTS user (using the
number 28761000) connected to an IP PBX makes a local call by dialing 0755 28961000. The
configured call route connects local outgoing calls with call prefix 2896 through an AT0 trunk.
Therefore, a pre-routing number change plan needs to be configured to remove 0755 from the
called number.
Pre-configuration Tasks
Before configuring the pre-routing number change service, complete the following tasks:
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Data Preparation
To configure the pre-routing number change service, you need the following data.
No.
Data
Enterprise, DN set, Centrex group, and call prefix bound to the pre-routing number
change plan
Calling numbers that need to be changed before routing, calling number change rule,
and called number change rule
(Optional) New enterprise name and Centrex group name used after number change
Procedure
Step 1 Run:
system-view
A calling number change rule and a called number change rule must be configured simultaneously. If the
calling number or called number keeps unchanged, set the number rule mode to no-change.
l Run:
caller { del-then-insert del-offset del-len insert-telnum | del del-offsetval
del-lenval | insert insert-offset insert-telnum-val | no-change }
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Usage Scenario
You can configure post-routing number change plans to define various dialing modes and change
the calling number displayed on the called party's phone. A post-routing number change plan
changes a called number to a long number. A post-routing number change plan can change a
called number to a long number ensure that it complies with the required number format. For
example, a POTS user (using the number 2876100) connected to an IP PBX makes a national
toll call by dialing 075528560982. A post-routing number change plan adds 12523 to the called
number 07552856098. 12523 is the call prefix defined by the carrier for the enterprise. When
the carrier's device detects the call prefix 12523, it connects the outgoing call through the
matching trunk. This reduces the call fees of the enterprise.
Pre-configuration Tasks
Before configuring the post-routing number change service, complete the following tasks:
l
Data Preparation
To configure the post-routing number change service, you need the following data.
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No.
Data
Procedure
Step 1 Run:
system-view
A calling number change rule and a called number change rule must be configured simultaneously. If the
calling number or called number keeps unchanged, set the number rule mode to no-change.
l Run:
caller { del-then-insert del-offset del-len insert-telnum | del del-offsetval
del-lenval | insert insert-offset insert-telnum-val | no-change }
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2 PBX Configuration
Usage Scenario
To configure the PBX line selection service, create a PBX group and allocate an access number
to all users in the PBX group. When a user on the public network dials the access number, the
call processing program selects an idle line to connect the call by using the configured line
selection mode.
Pre-configuration Tasks
Before configuring the PBX line selection service, complete the following tasks:
l
Creating an enterprise
Data Preparation
To configure the PBX line selection service, you need to the following data.
No.
Data
Procedure
Step 1 Run:
system-view
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The ringing duration, ringing mode, and line selection mode are configured.
----End
Usage Scenario
Before the co-group pickup service is configured, some calls will be missing if the called parties
do not answer the calls in a timely manner. After the co-group pickup service is configured,
users in the same group can answer calls for each other on their own phones. For example, if
user A does not answer a call, user B in the same group can dial the service access code plus
user A's phone number to answer the call. This service reduces missing calls.
Pre-configuration Tasks
Before configuring the co-group pickup service, complete the following tasks:
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Creating an enterprise
2 PBX Configuration
Data Preparation
To configure the co-group pickup service, you need the following data.
No.
Data
Procedure
Step 1 Run:
system-view
Before configuring the pickup service, run the service-right pickup-in-group enable command in the PBX
user view to enable the pickup service.
----End
Usage Scenario
Before simultaneous ringing is configured, some calls will be missing if the called parties do
not answer the calls in a timely manner. After simultaneous ringing is configured, when user B
calls user A using the access number of a simultaneous ringing group, all the idle member phones
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in the group ring simultaneously, and user A can answer the call using any ringing phone.
Simultaneous ringing reduces missing calls and improves the call connection ratio without
requiring additional devices.
Pre-configuration Tasks
Before configuring simultaneous ringing, complete the following tasks:
l
Creating an enterprise
Data Preparation
To configure simultaneous ringing, you need the following data.
No.
Data
Procedure
Step 1 Run:
system-view
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2 PBX Configuration
Usage Scenario
Before sequential ringing is configured, some calls will be missing if the called parties do not
answer the calls in a timely manner. After sequential ringing is configured, when user B calls
user A using the access number of a sequential ringing group, member phones in the group ring
in the configured sequence. Sequential ringing reduces missing calls and improves the call
connection ratio without requiring additional devices.
Pre-configuration Tasks
Before configuring sequential ringing, complete the following tasks:
l
Creating an enterprise
Data Preparation
To configure sequential ringing, you need the following data.
No.
Data
(Optional) Ringing duration and line selection mode of the sequential ringing group
Procedure
Step 1 Run:
system-view
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2 PBX Configuration
The ringing duration and line selection mode are set for the sequential ringing group.
----End
Usage Scenario
Before the ONLY service is configured, some calls will be missing if the called parties do not
answer the calls in a timely manner. After the ONLY service is configured, a user is bound to
multiple terminals. When user B calls user A using the access number of the ONLY service,
multiple terminals of user A ring according to the configured rules, and user A can select one
terminal to answer the call. Therefore, the ONLY service reduces missing calls and improves
the call connection ratio.
Pre-configuration Tasks
Before configuring the ONLY service, complete the following tasks:
l
Creating an enterprise
Data Preparation
To configure the ONLY service, you need the following data.
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No.
Data
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2 PBX Configuration
No.
Data
Enterprise and DN set bound to the ONLY service and access number of the ONLY
service
(Optional) Ringing duration, ringing mode, and line selection mode of the ONLY
service
Procedure
Step 1 Run:
system-view
The ringing duration, ringing mode, and line selection mode are set for the ONLY service.
----End
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2 PBX Configuration
Applicable Environment
After a trunk group is configured, all calls meeting the call conditions configured in the trunk
group are allowed. To restrict call rights of users, you can configure a rule set for the trunk group
to allow or reject calls of specified users.
Pre-configuration Tasks
Before configuring the blacklist or whitelist function, complete the following task:
l
Data Preparation
To configure the blacklist or whitelist function, you need the following data.
No.
Data
Procedure
Step 1 Run:
system-view
The rule set type (blacklist or whitelist) and restricted call type are specified.
Step 5 Run:
enterprise enterprise-name
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Applicable Environment
The DISA service can be used in the following scenarios:
l
PSTN
User A
Port 1/0/4
AT0 trunk
External number:
25600000
Internal number: 800
PBX
User B
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User C
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2 PBX Configuration
PSTN
User A
Port 1/0/4
AT0 trunk
PBX
User B
User C
Pre-configuration Tasks
Before configuring the DISA service, complete the following task:
l
Data Preparation
To configure the DISA service, you need the following data.
Issue 01 (2012-04-20)
No.
Data
Enterprise name
184
2 PBX Configuration
Procedure
Step 1 Create an account set and configure the authentication mode for the account set.
1.
Run:
system-view
Run:
voice
Run:
account-set account-set-name [account | password ]
Run:
enterprise enterprise-name
When the account set uses account authentication, run the account account-value
password { simple | cipher } password-value command to configure an account and
password for the account set.
When the account set uses password authentication, run the password { simple | cipher}
password-value command to configure a password for the account set.
Step 3 Run:
quit
Run:
disa disa-name [ enterprise enterprise-name ]
A DISA service profile is created and the DISA service view is displayed.
2.
Run:
dn-set dn-set-name
Use either of the following methods to configure the DISA service access number. (Use
the first method when the PBX needs to control outgoing calls from external users.)
l Run the access-telno [ country-code country-code-value ] [ area-code area-codevalue ] telno-value command to set to the DISA service access number to a user phone
number.
l Run the access-telno centrex centrex-name centrex-number command to set the DISA
service access number to a Centrex group number.
4.
Run:
use-account-set account-set-name
185
5.
2 PBX Configuration
(Optional) Run:
dst-enterprise dst-enterprise-name [ dst-dnset dst-dnset-name ] [ dst-centrex
dst-centrex-name ]
The destination enterprise, destination DN set, and destination Centrex are specified for
the DISA service.
----End
Run the display voice account-set [ account-value ] command to check the configuration
of the account set.
Run the display voice disa [ disa-name ] command to check the DISA service
configuration.
Applicable Environment
During a conversation between user A and user B, user A or user B can press the hookflash and
then dial 3 and user C's phone number to invite user C to the conversation. If they want to invite
other users to the conversation, the intermediate service is required. If user A has the intermediate
conference right, user A can press hookflash and dial 6 and user D's phone number to invite user
D to the conference.
Figure 2-14 shows the intermediate conference usage scenario.
Figure 2-14 Networking of the intermediate conference service
IMS/IP
network
PBX
User A
(POTS)
User C
(POTS)
Issue 01 (2012-04-20)
SIPAG
User D
(POTS)
User B
(SIPUE)
User E
(SIPUE)
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2 PBX Configuration
NOTE
To use the intermediate conference service, the user that initiates an intermediate conference must be in a
three-party conversation.
Pre-configuration Tasks
Before configuring the intermediate conference service, complete the following task:
l
Perform the following steps to configure the intermediate conference service for the POTS
and SIPUE users connected to the PBX:
Procedure
1.
Run:
system-view
Run:
voice
Run:
pbxuser name
Run:
service-right instant-conference enable
Run:
quit
Run:
callprefix callprefix-name
Run:
call-type category supplementary-service attribute 191
The call type and call attribute are configured for the intermediate conference service.
8.
Run:
quit
(Optional) Run:
pbx service-priority
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NOTE
l By default, the three-party service has a higher priority than the intermediate conference
service. The three-party and intermediate conference services have the same service process
when there are three parties in a conversation. When a user has the rights of both the threeparty service and intermediate conference service, the PBX determines whether to start the
three-party service or intermediate conference service based on priorities of the two
services.
l To configure the PBX to start the intermediate conference service, you can also run the
service-right three-party disable command to disable the three-party service. In this case,
you do not need to perform step 9.
Perform the following steps to enable the POTS and SIPUE users connected to the SIP AG
to use the intermediate conference service:
1.
Run:
system-view
Run:
voice
Run:
sipag
Run:
conference-factory-uri
Internal calls of the enterprises are connected through the PBX, and outgoing calls from
the enterprises are connected to external users through the AT0 trunk.
The carrier allocates the number 56623000 to the enterprise. External users can dial the
number 56623000 to query internal extension number. External users can also dial the
number 56623000, and then the call is transferred to an internal user.
NOTE
This example uses the voice tone "Please dial the extension number, or dial zero for the operator."
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Figure 2-15 Networking for configuring voice services for a small- or medium-sized enterprise
PSTN
AT0 trunk
Port 1/0/4
RouterA
Port 1/0/0
Port 1/0/2
Port 1/0/1
User A
User C
User B
Configuration Roadmap
The configuration roadmap is as follows:
1.
2.
3.
4.
Configure prefixes.
5.
6.
7.
8.
Data Preparation
To complete the configuration, you need the following data:
l
The country code and region code in China are used as an example.
Extension numbers of User A, User B, and User C: 800, 801, and 802 (the access number
of the IVR group is 800)
DN set: local
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2 PBX Configuration
IVR group name and user-defined RBT file name: ivr1 and sss.wav
NOTE
If the user-defined RBT is used, ensure that the RBT file has been made and uploaded/downloaded
to the storage media
Procedure
Step 1 Set the service mode to PBX.
NOTE
The PBX functions are controlled by the license. By default, PBX functions are disabled on a newly
purchased device. To use the PBX functions, apply for and purchase the license from the Huawei local
office.
<RouterA> system-view
[RouterA] voice
[RouterA-voice] service-mode pbx
Changing of the protocol configuration takes effect after you save the data and
then reboot the system. Are you sure to change the protocol configuration? (y/n
)[n] : y
[RouterA-voice] quit
[RouterA] quit
<RouterA> save
The current configuration will be written to the device.
Are you sure to continue? [y/n]y
<RouterA> reboot
Info: The system is now comparing the configuration, please wait.
System will reboot! Continue ? [y/n]:y
Info: system is rebooting ,please wait...
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pots enterprise hw
port 1/0/0
telno country-code 86 area-code 25 800
dn-set local
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2 PBX Configuration
# Configure User B.
[RouterA-voice] pbxuser 801
[RouterA-voice-pbxuser-801]
[RouterA-voice-pbxuser-801]
[RouterA-voice-pbxuser-801]
[RouterA-voice-pbxuser-801]
toll
[RouterA-voice-pbxuser-801]
pots enterprise hw
port 1/0/1
telno country-code 86 area-code 25 801
dn-set local
call-right in international-toll out internationalquit
# Configure User C.
[RouterA-voice] pbxuser 802
[RouterA-voice-pbxuser-802]
[RouterA-voice-pbxuser-802]
[RouterA-voice-pbxuser-802]
[RouterA-voice-pbxuser-802]
toll
[RouterA-voice-pbxuser-802]
pots enterprise hw
port 1/0/2
telno country-code 86 area-code 25 802
dn-set local
call-right in international-toll out internationalquit
callprefix 9
trunk-group at0
caller no-change
called del 7 1
quit
When external users dial the number 56623000, they can dial extension numbers to
communicate with internal users.
2.
3.
----End
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Configuration Files
Configuration file of RouterA
voice
pbx default-area-code 25
#
enterprise hw
crbt-file flash:/sss.wav status pass
dn-set local
#
r2 signalling-type argentina
#
r2 signalling-type brazil
#
r2 signalling-type mexico
#
r2 signalling-type standard
#
trunk-group at0 fxo
enterprise hw dn-set local
call-right in international-toll out international-toll
trunk-at0 1/0/4 default-called-telno 800 reversepole-detect disable
#
callprefix 8
enterprise hw dn-set local
prefix 8
call-type category basic-service attribute 0
digit-length 3 4
destination-location inter-office
#
callprefix 9
enterprise hw dn-set local
prefix 9
call-type category basic-service attribute 0
digit-length 1 15
destination-location inter-office
callroute trunkgroup1 at0
#
pbxuser 800 pots enterprise hw
port 1/0/0
telno 800
dn-set local
call-right in international-toll out international-toll
#
pbxuser 801 pots enterprise hw
port 1/0/1
telno 801
dn-set local
call-right in international-toll out international-toll
#
pbxuser 802 pots enterprise hw
port 1/0/2
telno 802
dn-set local
call-right in international-toll out international-toll
#
pbxusergroup ivr1 ivr enterprise hw
dn-set local
access-telno 800
console-telno 2
tone-id file flash:/sss.wav
destination dn-set DefaultDialPlan
group-member pbxuser 800 member-index 1
#
afterroute-change 9
callprefix 9
trunk-group at0
caller no-change
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called del 7 1
#
return
The carrier allocates the number 56623000 to enterprise A. If external users dial the number
56623000, the phone of User A rings and the call transfer service is enabled. When external
users call other internal users, the phone of User A transfers the calls.
The carrier allocates the number 56623001 to enterprise B. If external users dial the number
56623001, the phone of User C rings and the call transfer service is enabled. When external
users call other internal users, the phone of User C transfers the calls.
Figure 2-16 Networking for configuring the AR to implement communication for different
enterprises
IMS/IP
network
Trunk
Router Eth2/0/0
Port 1/0/0
User A
Port
1/0/1
Port 1/0/3
Port
1/0/2
User B
User D
User C
Enterprise A
Enterprise B
Campus network
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Configuration Roadmap
The configuration roadmap is as follows:
1.
2.
3.
4.
5.
6.
Configure prefixes.
7.
8.
9.
Data Preparation
To complete the configuration, you need the following data:
l
The country code and region code in China are used as an example.
Numbers of User A, User B, User C, and User D: 2000, 2001, 3000, and 3001
Enterprise huawei to which user A belongs, DN set local, intra-office call prefix 2, and
inter-office call prefix 8
Enterprise huawei to which user B belongs, DN set local1, intra-office call prefix 3, and
inter-office call prefix 9
Procedure
Step 1 Set the service mode to PBX.
NOTE
The PBX functions are controlled by the license. By default, PBX functions are disabled on a newly
purchased device. To use the PBX functions, apply for and purchase the license from the Huawei local
office.
<Huawei> system-view
[Huawei] voice
[Huawei-voice] service-mode pbx
Changing of the protocol configuration takes effect after you save the data and
then reboot the system. Are you sure to change the protocol configuration? (y/n
)[n] : y
[Huawei-voice] quit
[Huawei] quit
<Huawei> save
The current configuration will be written to the device.
Are you sure to continue? [y/n]y
<Huawei> reboot
Info: The system is now comparing the configuration, please wait.
System will reboot! Continue ? [y/n]:y
Info: system is rebooting ,please wait...
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pots enterprise hw
port 1/0/0
telno country-code 86 area-code 25 2000
dn-set local
call-right in international-toll out internationalservice-right call-transfer enable
quit
# Configure User B.
[Huawei-voice] pbxuser 2001
[Huawei-voice-pbxuser-2001]
[Huawei-voice-pbxuser-2001]
[Huawei-voice-pbxuser-2001]
[Huawei-voice-pbxuser-2001]
toll
[Huawei-voice-pbxuser-2001]
pots enterprise hw
port 1/0/1
telno country-code 86 area-code 25 2001
dn-set local
call-right in international-toll out internationalquit
# Configure User C.
[Huawei-voice] pbxuser 3000
[Huawei-voice-pbxuser-3000]
[Huawei-voice-pbxuser-3000]
[Huawei-voice-pbxuser-3000]
[Huawei-voice-pbxuser-3000]
toll
[Huawei-voice-pbxuser-3000]
[Huawei-voice-pbxuser-3000]
# Configure User D.
[Huawei-voice] pbxuser 3001
[Huawei-voice-pbxuser-3001]
[Huawei-voice-pbxuser-3001]
[Huawei-voice-pbxuser-3001]
[Huawei-voice-pbxuser-3001]
toll
[Huawei-voice-pbxuser-3001]
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[Huawei-voice-trunkgroup-sipat01]
huawei.com
[Huawei-voice-trunkgroup-sipat01]
[Huawei-voice-trunkgroup-sipat01]
3000
[Huawei-voice-trunkgroup-sipat01]
Note: Trunkgroup reset succeeds.
[Huawei-voice-trunkgroup-sipat01]
home-domain
register-id 56623001
trunk-sipat0 56623001 default-called-telno
reset
quit
callprefix 8
trunk-group sipat0
caller no-change
called del 7 1
quit
callprefix 9
trunk-group sipat01
caller no-change
called del 7 1
quit
2.
3.
4.
When dialing the number 56623000, external users can talk with User A and User B.
5.
When dialing the number 56623001, external users can talk with User C and User D.
----End
Configuration Files
Configuration file of the router
#
voice
voip-address signalling interface Ethernet 0/0/1 192.168.1.3
voip-address media interface Ethernet 0/0/1 192.168.1.3
pbx default-area-code 25
#
dspattribute
#
enterprise hw
dn-set local
#
enterprise hw1
dn-set local1
#
sipserver
signalling-address ip 192.168.1.3 port 5060
media-ip 192.168.1.3
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register-uri huawei.com
home-domain huawei.com
#
r2 signalling-type argentina
#
r2 signalling-type brazil
#
r2 signalling-type mexico
#
r2 signalling-type standard
#
trunk-group sipat0 sip trunk-circuit
enterprise hw dn-set local
call-right in international-toll out international-toll
default-caller-telno 2000
signalling-address ip 192.168.1.3 port 5070
media-ip 192.168.1.3
peer-address static 192.168.1.1 5060
register-uri huawei.com
home-domain huawei.com
register-id 56623000
trunk-sipat0 56623000 default-called-telno 2000
#
trunk-group sipat01 sip trunk-circuit
enterprise hw1 dn-set local1
call-right in international-toll out international-toll
default-caller-telno 3000
signalling-address ip 192.168.1.3 port 5080
media-ip 192.168.1.3
peer-address static 192.168.1.1 5060
register-uri huawei.com
home-domain huawei.com
register-id 56623001
trunk-sipat0 56623001 default-called-telno 3000
#
callprefix 2
enterprise hw dn-set local
prefix 2
call-type category basic-service attribute 0
digit-length 4 8
#
callprefix 3
enterprise hw1 dn-set local1
prefix 3
call-type category basic-service attribute 0
digit-length 4 8
#
callprefix 8
enterprise hw dn-set local
prefix 8
call-type category basic-service attribute 0
digit-length 1 15
destination-location inter-office
callroute trunkgroup1 sipat0
#
callprefix 9
enterprise hw1 dn-set local1
prefix 9
call-type category basic-service attribute 0
digit-length 1 15
destination-location inter-office
callroute trunkgroup1 sipat01
#
pbxuser 2000 pots enterprise hw
port 1/0/0
telno 2000
dn-set local
call-right in international-toll out international-toll
#
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afterroute-change 8
callprefix 8
trunk-group sipat0
caller no-change
called del 7 1
#
afterroute-change 9
callprefix 9
trunk-group sipat01
caller no-change
called del 7 1
#
return
The carrier allocates the number 56623000 to the enterprise headquarters. If external users
dial the number 56623000, the phone of User A rings and the call transfer service is enabled.
When external users call other internal users, the phone of User A transfers the calls.
The carrier allocates the number 28963000 to the enterprise branch. If external users dial
the number 28963000, the phone of User C rings and the call transfer service is enabled.
When external users call other internal users, the phone of User C transfers the calls.
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Figure 2-17 Networking diagram for configuring calls between the headquarters and branch
Enterprise A
Branch
User C
User D
Port1/0/0
Router B
Port1/0/1
Port1/0/4
Eth2/0/0
PSTN
H323 trunk
IP network
PSTN
H323 trunk
Eth2/0/0
Port1/0/4
Router A
Port1/0/0
Port1/0/1
User A
User B
Enterprise A
Headquarters
Configuration Roadmap
The configuration roadmap is as follows:
1.
2.
3.
4.
5.
6.
Configure prefixes.
7.
8.
9.
Data Preparation
To complete the configuration, you need the following data:
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Country code 86, area code 25 of Router A, and area code 755 of Router B
NOTE
The country code and region code in China are used as an example. During deployment, configure
the country code and region code based on actual networking.
Enterprise hw to which user A and user B belong, DN set local, call prefix 2222, interoffice prefix 9 of the AT0 trunk, and inter-office prefix 20000 between the headquarters
and branch
Enterprise hw to which user C and user D belong, DN set local, call prefix 3333, interoffice prefix 9 of the AT0 trunk, and inter-office prefix 20000 between the headquarters
and branch
Procedure
Step 1 Set the service mode to PBX on RouterA and RouterB.
NOTE
The PBX functions are controlled by the license. By default, PBX functions are disabled on a newly
purchased device. To use the PBX functions, apply for and purchase the license from the Huawei local
office.
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# Configure RouterB.
[RouterB] voice
[RouterB-voice] pbx default-country-code 86 default-area-code 755
# Configure RouterB.
[RouterB-voice] enterprise hw
[RouterB-voice-enterprise-hw] dn-set local
[RouterB-voice-enterprise-hw] quit
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[RouterB-voice] h323-attribute
[RouterB-voice-h323-attribute] localip 192.168.1.2
[RouterB-voice-h323-attribute] reset
H323 system parameters reset successfully!
[RouterA-voice-h323-attribute] quit
pots enterprise hw
port 1/0/0
telno country-code 86 area-code 25 22223000
dn-set local
call-right in international-toll out
service-right call-transfer enable
quit
# Configure user B.
[RouterA-voice] pbxuser 22223001 pots enterprise hw
[RouterA-voice-pbxuser-22223001] port 1/0/1
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telno country-code 86 area-code 25 22223001
dn-set local
call-right in international-toll out
quit
# Configure user C.
[RouterB-voice] pbxuser 33333000
[RouterB-voice-pbxuser-33333000]
[RouterB-voice-pbxuser-33333000]
[RouterB-voice-pbxuser-33333000]
[RouterB-voice-pbxuser-33333000]
international-toll
[RouterB-voice-pbxuser-33333000]
[RouterB-voice-pbxuser-33333000]
pots enterprise hw
port 1/0/0
telno country-code 86 area-code 755 33333000
dn-set local
call-right in international-toll out
service-right call-transfer enable
quit
# Configure user D.
[RouterB-voice] pbxuser 33333001
[RouterB-voice-pbxuser-33333001]
[RouterB-voice-pbxuser-33333001]
[RouterB-voice-pbxuser-33333001]
[RouterB-voice-pbxuser-33333001]
international-toll
[RouterB-voice-pbxuser-33333000]
pots enterprise hw
port 1/0/1
telno country-code 86 area-code 755 33333001
dn-set local
call-right in international-toll out
quit
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2 PBX Configuration
2.
3.
4.
When dialing the number 56623000, external users can talk with user A and user B.
5.
When dialing the number 28963000, external users can talk with user C and user D.
----End
Configuration Files
# Configuration file of RouterA
#
voice
voip-address signalling interface Ethernet 0/0/1 192.168.1.1
voip-address media interface Ethernet 0/0/1 192.168.1.1
sip-reg-count-per-second 4294967295
pbx default-area-code 25
#
h323-
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attribute
localip 192.168.1.1
#
enterprise hw
dn-set local
#
r2 signalling-type argentina
#
r2 signalling-type brazil
#
r2 signalling-type mexico
#
r2 signalling-type standard
#
trunk-group at0 fxo
enterprise hw dn-set local
call-right in international-toll out international-toll
trunk-at0 1/0/4 default-called-telno 22223000 reversepole-detect disable
#
trunk-group h323 h323 symmetrical
enterprise hw dn-set local
call-right in international-toll out international-toll
signalling-ip ip 192.168.1.1
media-ip 192.168.1.1
peer-address static 192.168.1.2 1720
#
callprefix 9
enterprise hw dn-set local
prefix 9
call-type category basic-service attribute 0
digit-length 1 15
destination-location inter-office
callroute trunkgroup1 at0
#
callprefix 2222
enterprise hw dn-set local
prefix 2222
call-type category basic-service attribute 0
digit-length 8 9
#
callprefix 20000
enterprise hw dn-set local
prefix 20000
call-type category basic-service attribute 0
digit-length 5 20
destination-location inter-office
callroute trunkgroup1 h323
#
pbxuser 22223000 pots enterprise hw
port 1/0/0
telno 22223000
dn-set local
call-right in international-toll out international-toll
service-right call-transfer enable
#
pbxuser 22223001 pots enterprise hw
port 1/0/1
telno 22223001
dn-set local
call-right in international-toll out international-toll
#
afterroute-change 9
callprefix 9
trunk-group at0
caller no-change
called del 7 1
#
afterroute-change 20000
callprefix 20000
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trunk-group h323
caller no-change
called del 7 5
#
return
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port 1/0/1
telno 33333001
dn-set local
call-right in international-toll out international-toll
#
afterroute-change 9
callprefix 9
trunk-group at0
caller no-change
called del 8 1
#
afterroute-change 20000
callprefix 20000
trunk-group h323
caller no-change
called del 8 5
#
return
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Figure 2-18 Networking diagram for configuring calls between the headquarters and branch
Enterprise A
Branch
User C
User D
Port1/0/0
Router B
H323 trunk
Port1/0/1
Eth2/0/0
IP network
Eth2/0/0
H323 trunk
Router A
Port1/0/0
Port1/0/1
User A
User B
Enterprise A
Headquarters
Configuration Roadmap
The configuration roadmap is as follows:
1.
2.
3.
4.
5.
6.
Configure prefixes.
7.
8.
9.
Data Preparation
To complete the configuration, you need the following data (data of Router A):
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The country code and region code in China are used as an example. During deployment, configure
the country code and region code based on actual networking.
Number of user A: 22223000 and number of user B: 22223001 Number of user C: 33333000
and number of user D: 33333001
Signaling and media IP addresses and signaling port number of the headquarters:
192.168.1.1 Signaling and media IP addresses and signaling port number of the branch:
192.168.1.2
Enterprise huawei to which user A and user B belong, DN set local, call prefix 2222, and
inter-office prefix 20000 between the headquarters and branch
Procedure
Step 1 Configure RouterA to work in PBX mode.
NOTE
l The PBX functions are controlled by the license. By default, PBX functions are disabled on a newly
purchased device. To use the PBX functions, apply for and purchase the license from the Huawei local
office.
l The preceding configuration is the configuration of RouterA.
210
2 PBX Configuration
# Configure RouterA.
[RouterA-voice] enterprise hw
[RouterA-voice-enterprise-hw] dn-set local
[RouterA-voice-enterprise-hw] quit
pots enterprise hw
port 1/0/0
telno country-code 86 area-code 25 22223000
dn-set local
call-right in international-toll out
service-right call-transfer enable
quit
# Configure user B.
[RouterA-voice] pbxuser 22223001
[RouterA-voice-pbxuser-22223001]
[RouterA-voice-pbxuser-22223001]
[RouterA-voice-pbxuser-22223001]
[RouterA-voice-pbxuser-22223001]
international-toll
[RouterA-voice-pbxuser-22223001]
pots enterprise hw
port 1/0/1
telno country-code 86 area-code 25 22223001
dn-set local
call-right in international-toll out
quit
The same gateway ID must be configured on devices of the headquarters and branch.
[RouterA-voice] trunk-group h323 h323 register-gateway
[RouterA-voice-trunkgroup-h323] enterprise hw dn-set local
[RouterA-voice-trunkgroup-h323] call-right in international-toll out
international-toll
[RouterA-voice-trunkgroup-h323] gwid h323
[RouterA-voice-trunkgroup-h323] signalling-ip ip 192.168.1.1
[RouterA-voice-trunkgroup-h323] media-ip 192.168.1.1
[RouterA-voice-trunkgroup-h323] quit
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2 PBX Configuration
callprefix 20000
trunk-group h323
caller no-change
called del 7 5
2.
----End
Configuration Files
# Configuration file of RouterA
#
voice
voip-address signalling interface GigabitEthernet 0/0/1 192.168.1.1
voip-address media interface GigabitEthernet 0/0/1 192.168.1.1
sip-reg-count-per-second 4294967295
pbx default-area-code 25
#
h323attribute
localip 192.168.1.1
#
enterprise hw
dn-set local
#
r2 signalling-type argentina
#
r2 signalling-type brazil
#
r2 signalling-type mexico
#
r2 signalling-type standard
#
trunk-group h323 h323 register-gateway
enterprise hw dn-set local
call-right in international-toll out international-toll
signalling-ip ip 192.168.1.1
media-ip 192.168.1.1
gwid h323
#
callprefix 2222
enterprise hw dn-set local
prefix 2222
call-type category basic-service attribute 0
digit-length 8 9
#
callprefix 20000
enterprise hw dn-set local
prefix 20000
call-type category basic-service attribute 0
digit-length 5 20
destination-location inter-office
callroute trunkgroup1 h323
#
pbxuser 22223000 pots enterprise hw
port 1/0/0
telno 22223000
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dn-set local
call-right in international-toll out international-toll
service-right call-transfer enable
#
pbxuser 22223001 pots enterprise hw
port 1/0/1
telno 22223001
dn-set local
call-right in international-toll out international-toll
#
afterroute-change 20000
callprefix 20000
trunk-group h323
caller no-change
called del 7 5
#
return
2.16.5 Example for Using the SIP Trunk Group to Configure Calls
Between the Headquarters and Branch
Networking Requirements
As shown in Figure 2-19, the headquarters and branch of enterprise A are located in different
areas. RouterA and RouterB use SIP trunks to connect the headquarters and branch. After voice
services are deployed on RouterA and RouterB, enterprise users can use the voice services across
areas.Internal users call external users through the AT0 trunk. The requirements are as follows:
l
The carrier allocates the number 56623000 to the enterprise headquarters. If external users
dial the number 56623000, the phone of User A rings and the call transfer service is enabled.
When external users call other internal users, the phone of User A transfers the calls.
The carrier allocates the number 28963000 to the enterprise branch. If external users dial
the number 28963000, the phone of User C rings and the call transfer service is enabled.
When external users call other internal users, the phone of User C transfers the calls.
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Figure 2-19 Networking for configuring communication between the headquarters and branch
Enterprise A
Branch
User C
User D
Port1/0/0
Port1/0/1
Router B
Port1/0/4
Eth2/0/0
SIP trunk
PSTN
IP network
PSTN
SIP trunk
Port1/0/4
Eth2/0/0
Router A
Port1/0/0
Port1/0/1
User A
User B
Enterprise A
Headquarters
Configuration Roadmap
The configuration roadmap is as follows:
1.
2.
3.
4.
5.
6.
Configure prefixes.
7.
8.
9.
Data Preparation
To complete the configuration, you need the following data:
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Country code 86, area code 2 of Router A, and area code 755 of Router B
NOTE
The country code and region code in China are used as an example.
Internal numbers of User A, User B, User C, and User D: 22223000, 22223001, 33333000,
and 33333001
Signaling and media IP addresses and signaling port number of the headquarters:
192.168.1.1 and 5070
Signaling and media IP addresses and signaling port number of the branch: 192.168.1.2
and 5070
Enterprise huawei to which user A and user B belong, DN set local, call prefix 2222, interoffice call prefix 9 of the AT0 trunk, and inter-office call prefix 20000 between the
headquarters and branch
Enterprise huawei to which user C and user D belong, DN set local, call prefix 3333, interoffice call prefix 9 of the AT0 trunk, and inter-office call prefix 20000 between the
headquarters and branch
Procedure
Step 1 Set the service mode to PBX on RouterA and RouterB.
NOTE
The PBX functions are controlled by the license. By default, PBX functions are disabled on a newly
purchased device. To use the PBX functions, apply for and purchase the license from the Huawei local
office.
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2 PBX Configuration
# Configure RouterB.
[RouterB-voice] pbx
[RouterB-voice] enterprise hw
[RouterB-voice-enterprise-hw] dn-set local
[RouterB-voice-enterprise-hw] quit
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2 PBX Configuration
[RouterB-voice] sipserver
[RouterB-voice-sipserver] signalling-address ip 192.168.1.2 port 5060
[RouterB-voice-sipserver] media-ip 192.168.1.2
[RouterB-voice-sipserver] register-uri huawei.com
[RouterB-voice-sipserver] home-domain huawei.com
[RouterB-voice-sipserver] reset
SIP server reset succeeds.
[RouterB-voice-sipserver] quit
Issue 01 (2012-04-20)
pots enterprise hw
port 1/0/0
telno country-code 86 area-code 25 22223000
dn-set local
call-right in international-toll out
service-right call-transfer enable
quit
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2 PBX Configuration
# Configure User B.
[RouterA-voice] pbxuser 22223001
[RouterA-voice-pbxuser-22223001]
[RouterA-voice-pbxuser-22223001]
[RouterA-voice-pbxuser-22223001]
[RouterA-voice-pbxuser-22223001]
international-toll
[RouterA-voice-pbxuser-22223001]
pots enterprise hw
port 1/0/1
telno country-code 86 area-code 25 22223001
dn-set local
call-right in international-toll out
quit
# Configure User C.
[RouterB-voice] pbxuser 33333000
[RouterB-voice-pbxuser-33333000]
[RouterB-voice-pbxuser-33333000]
[RouterB-voice-pbxuser-33333000]
[RouterB-voice-pbxuser-33333000]
international-toll
[RouterB-voice-pbxuser-33333000]
[RouterB-voice-pbxuser-22223001]
pots enterprise hw
port 1/0/0
telno country-code 86 area-code 755 33333000
dn-set local
call-right in international-toll out
service-right call-transfer enable
quit
# Configure User D.
[RouterB-voice] pbxuser 33333001
[RouterB-voice-pbxuser-33333001]
[RouterB-voice-pbxuser-33333001]
[RouterB-voice-pbxuser-33333001]
[RouterB-voice-pbxuser-33333001]
international-toll
[RouterB-voice-pbxuser-22223001]
pots enterprise hw
port 1/0/1
telno country-code 86 area-code 755 33333001
dn-set local
call-right in international-toll out
quit
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2.
3.
4.
When dialing the number 56623000, external users can talk with User A and User B.
5.
When dialing the number 28963000, external users can talk with User C and User D.
----End
Issue 01 (2012-04-20)
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2 PBX Configuration
Configuration Files
# Configuration file of RouterA
voice
voip-address media interface Ethernet 2/0/0 192.168.1.1
voip-address signalling interface Ethernet 2/0/0 192.168.1.1
pbx default-area-code 25
#
dspattribute
#
enterprise hw
dn-set local
#
sipserver
signalling-address ip 192.168.1.1 port 5060
media-ip 192.168.1.1
register-uri huawei.com
home-domain huawei.com
#
r2 signalling-type argentina
#
r2 signalling-type brazil
#
r2 signalling-type mexico
#
r2 signalling-type standard
#
trunk-group at0 fxo
enterprise hw dn-set local
call-right in international-toll out international-toll
trunk-at0 1/0/4 default-called-telno 22223000 reversepole-detect disable
#
trunk-group sipip sip no-register
enterprise hw dn-set local
call-right in international-toll out international-toll
signalling-address ip 192.168.1.1 port 5070
media-ip 192.168.1.1
peer-address static 192.168.1.2 5070
register-uri huawei.com
home-domain huawei.com
#
callprefix 9
enterprise hw dn-set local
prefix 9
call-type category basic-service attribute 0
digit-length 1 15
destination-location inter-office
callroute trunkgroup1 at0
#
callprefix 2222
enterprise hw dn-set local
prefix 2222
call-type category basic-service attribute 0
digit-length 8 9
#
callprefix 20000
enterprise hw dn-set local
prefix 20000
call-type category basic-service attribute 0
digit-length 5 20
destination-location inter-office
callroute trunkgroup1 sipip
#
pbxuser 22223000 pots enterprise hw
port 1/0/0
telno 22223000
dn-set local
call-right in international-toll out international-toll
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2 PBX Configuration
#
pbxuser 22223001 pots enterprise hw
port 1/0/1
telno 22223001
dn-set local
call-right in international-toll out international-toll
#
afterroute-change 9
callprefix 9
trunk-group at0
caller no-change
called del 7 1
#
afterroute-change 20000
callprefix 20000
trunk-group sipip
caller no-change
called del 7 5
#
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2 PBX Configuration
#
callprefix 20000
enterprise hw dn-set local
prefix 20000
call-type category basic-service attribute 0
destination-location inter-office
callroute trunkgroup1 sipip
#
pbxuser 33333000 pots enterprise hw
port 1/0/0
telno 33333000
dn-set local
call-right in international-toll out international-toll
service-right call-transfer enable
#
pbxuser 33333001 pots enterprise hw
port 1/0/1
telno 33333001
dn-set local
call-right in international-toll out international-toll
#
afterroute-change 9
callprefix 9
trunk-group at0
caller no-change
called del 8 1
#
afterroute-change 20000
callprefix
20000
trunk-group at0
caller no-change
called del 8 5
#
return
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