Professional Documents
Culture Documents
V200R002C01
Issue 01
Date 2012-04-20
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Intended Audience
This document provides the basic concepts, configuration procedures, and configuration
examples in different application scenarios of the voice feature.
Symbol Conventions
The symbols that may be found in this document are defined as follows.
Symbol Description
Command Conventions
The command conventions that may be found in this document are defined as follows.
Convention Description
&<1-n> The parameter before the & sign can be repeated 1 to n times.
Change History
Changes between document issues are cumulative. Therefore, the latest document issue contains
all updates made in previous issues.
Contents
2 PBX Configuration......................................................................................................................66
1 SIPAG Configuration
This chapter describes the concepts and configuration of SIPAG, and provides a configuration
example.
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.
l Among the AR1200 series routers, only the AR1220Vs and AR1220VWs support the voice features.
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 Eth1/0/0
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.
Basic voice service The basic voice service is the basic call connection Yes
function, including intra-office calls, local calls,
national toll calls, international toll calls, and transit
calls.
Three-party service The third-party service allows a calling party or called Yes
party in a conversation to call a third party without
ending the current conversation. Then the calling party
or original called party can implement a three-party
conversation or talk to the other two parties.
Call waiting When user A is talking with user B over the phone and Yes
service at this moment user C is calling user A, user A hears a
call waiting tone, indicating that there is a call waiting
for user A.
MWI service The message waiting indicator (MWI) service allows a Yes
user to read unread messages or leave messages. When
the called user is busy, the MWI is on, indicating that
there are leave messages.
Malicious call The user that registers the MCID service with the carrier Yes
identification can query the phone number of the attacker that initiates
(MCID) service malicious calls after performing relevant operations.
Call transfer The call transfer service allows the called party to Yes
service transfer an incoming call to a third party by pressing the
hookflash so that the calling party establishes a
connection with a new called party.
Call conference The call conference service allows more than three Yes
service parties to communicate together.
Calling line The CLIP service displays the calling number in onhook No
identification state or offhook state (for call waiting). The displayed
presentation information includes the phone number, name, date, and
(CLIP) service time.
Distinctive ringing The distinctive ringing service plays different ring tones No
service for incoming calls.
Advice of charge The AoC service enables the SIP AG to display the No
(AoC) service charge rate, fee notification during a call, and the total
fee of the call.
Urgent call process If the SIP AG finds an urgent call, the SIP AG inserts No
the urgent call flag into the SIP message.
Anonymous call The anonymous call service enables the called party not No
service to view information about incoming calls.
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:
To use the BEST function, you must purchase the CM&BEST License in addition to the precedng
licenses.
Applicable Environment
You can configure the AR1200 to work in SIP AG or PBX mode. Before configuring SIP AG
service features, configure the AR1200 to work in SIP AG mode. You can run the display voice
service-mode command to view the working mode of the AR1200. If the AR1200 works in
PBX mode, delete the PBX configurations and configure the AR1200 to work in SIP AG mode.
If the AR1200 works in SIP AG mode, skip this configuration.
Pre-configuration Tasks
Before configuring the AR1200 to work in SIP AG mode, complete the following task:
l Configuring IP addresses for interfaces and routing protocols to ensure connectivity
Procedure
Step 1 Run:
system-view
Step 2 Run:
voice
Step 3 Run:
service-mode sipag
Step 4 Run:
quit
NOTE
After the AR1200 is configured to work in SIP AG mode, restart the AR1200 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 Configuring the AR1200 to work in SIP AG mode
Data Preparation
To configure a SIP AG interface, you need the following data.
No. Data
2 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
----End
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
Step 2 Run:
voice
Step 4 Run:
sipag sipag-interface-id
----End
Procedure
l Run the display voice sipag [ sipag-interface-id { running | config } ] command to check
the configuration of SIP AG parameters.
l Run the display voice voip-address command to check the signaling and media IP
addresses pool.
----End
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 AR1200, including physical
attributes, electrical attributes, and KC attributes.
Pre-configuration Tasks
Before setting parameters for an FXS interface, complete the following task:
Data Preparation
To set parameters for an FXS interface, you need the following data.
No. Data
1 Polarity reversal pulse level width, polarity reversal mode, and dialing mode
3 High-level pulse width, low-level pulse width, KC accounting mode, and voltage
operating
Procedure
Step 1 Run:
system-view
Step 2 Run:
voice
Step 3 Run:
port fxs slotid/subcardid/portid
kc voltage voltage
----End
Applicable Environment
A basic rate access (BRA) interface connects to an ISDN phone. On the AR1200, 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 Configuring the AR1200 to work in SIPAG mode
l Ensuring that the 2BST board is working properly
Data Preparation
To set parameters for a BRA interface, you need the following data.
No. Data
1 Interface working mode, automatic deactivation delay, and Layer 1 activation mode
Procedure
Step 1 Run:
system-view
Step 5 Run:
work-mode { p2p | p2mp }
Step 6 Run:
remote-power enable
Step 7 Run:
auto-deactive enable
Step 8 Run:
auto-deactive delay delay
Step 9 Run:
active-mode { unstable | stable }
Step 10 Run:
alarm enable
----End
Applicable Environment
A PRA interface connects to a PBX or PSTN network. On the AR1200, 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:
Data Preparation
To set parameters for a PRA interface, you need the following data.
No. Data
Procedure
Step 1 Run:
system-view
----End
Applicable Environment
The AR1200 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 Configuring the AR1200 to work in SIP AG mode
l Configuring IP addresses and routing protocols for interfaces to ensure connectivity
Data Preparation
To set system parameters, you need the following data.
No. Data
1 Country/Region identifier
3 MWI mode
5 AC amplitude of the ringing current, frequency of the ringing current, and cadence
ratio
No. Data
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
Step 2 Run:
voice
Step 3 Run:
country { brazil | britain-msfuk | britain-etsi | bulgaria | china-hongkong |
china-mainland | egypt | france | singapore | thailand }
----End
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.
l 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.
Procedure
Step 1 Run:
system-view
Step 2 Run:
voice
Step 3 Run:
flash-hook lower lower-value
Step 4 Run:
flash-hook upper upper-value
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
----End
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
Step 3 Run:
pcm { a-law | u-law }
----End
Procedure
Step 1 Run:
system-view
Step 2 Run:
voice
----End
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 AR1200 to meet local standards.
Procedure
Step 1 Run:
system-view
Step 2 Run:
voice
----End
Procedure
Step 1 Run:
system-view
----End
Context
When the CLIP service is registered, CLIP parameters in offhook state need to be configured
on the AR1200 so that the AR1200 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
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
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 AR1200 so that the AR1200 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
Step 2 Run:
voice
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
The number of bits of the FSK synchronization mask in onhook state is set.
----End
Context
The AR1200 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
Step 2 Run:
voice
The AR1200 is enabled to restrict calls when the uplink bandwidth is enabled.
l Run:
media-bandwidth-control maximum max-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
Step 1 Run:
system-view
----End
Procedure
l Run the display voice configuration command to check the voice configuration.
l Run the display voice user-defined-ring [ring-index ] command to check user-defined
ring information.
l Run the display voice clip command to check CLIP parameters.
----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 Configuring the AR1200 to work in SIP AG mode
l Configuring IP addresses and routing protocols for interfaces to ensure connectivity
Data Preparation
To set SIP protocol stack parameters, you need the following data.
No. Data
Procedure
Step 1 Run:
system-view
Step 2 Run:
voice
Step 3 Run:
sip
----End
Procedure
Step 1 Run:
system-view
Step 2 Run:
voice
Step 3 Run:
sip
----End
Procedure
l Run the display voice sip command to check SIP parameters.
----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 Configuring the AR1200 to work in SIP AG mode
l Configuring IP addresses and routing protocols for interfaces to ensure connectivity
Data Preparation
To set DSP parameters, you need the following data.
No. Data
2 Default DSP channel code type and default interval at which the DSP channel
packetizes RTP packets
3 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
4 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
5 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
6 Interval at which a DSP channel sends RTCP packets and threshold for the number
of severe degrade seconds
8 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
----End
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
----End
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
----End
Procedure
Step 1 Run:
system-view
Step 4 Run:
autovbd { auto | host-controlled }
NOTE
By default, the VBD switching mode is host-controlled.
----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
Step 2 Run:
voice
Step 3 Run:
dsp-attribute
Step 4 Run:
codec code { g711a | g711u | g723_1 | g729 | g726-16k | g726-24k | g726-32k |
g726-40k } [ rtp-interval { 5ms | 10ms | 20ms | 30ms } ]
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
Step 2 Run:
voice
Step 3 Run:
dsp-attribute
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
----End
Procedure
Step 1 Run:
system-view
Step 2 Run:
voice
Step 3 Run:
dsp-attribute
----End
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
The RTP Control Protocol Extended Reports (RTCP XR) function is enabled.
l Run:
rtcp sev-degradethreshold sev-degradethresholdval
----End
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
Step 2 Run:
voice
Step 3 Run:
dsp-attribute
----End
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
Step 2 Run:
voice
Step 3 Run:
dsp-attribute
Step 4 Run:
template name
----End
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.
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
Step 2 Run:
voice
Step 3 Run:
dsp slot/dsp-index
----End
Procedure
l Run the display voice dsp-attribute command to check the DSP configuration.
l Run the display voice dsp state { slot/dsp-index | channel slot/dsp-index/channel }
command to check the status of a DSP or DSP channel.
l Run the display voice dsp-template command to check the DSP template configuration.
----End
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 Setting the operation mode to SIP AG
l Configuring a SIP AG interface
Data Preparation
To configure a SIP AG user, you need the following data.
No. Data
1 SIP AG user port number, SIP AG ID for the SIP AG user, and phone number of the
SIP AG user
2 (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
After creating a SIP AG user, set parameters for the SIP AG user, including the interface number
and basic phone number associated to the SIP AG User.
Step 4 Run:
agid sipag-interface-ID
Step 5 Run:
base-telno telno-value [ sipagusergroup usergroup-id ]
----End
Procedure
Step 1 Run:
system-view
Step 2 Run:
voice
Step 3 Run:
sipagusergroup sipag-interface-id usergroup-id
The mode used to manage users in the SIP AP user group is configured.
l Run:
register-uri-mode { inneruser | alone }
A URI is configured.
l Run:
endservice
----End
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
Step 2 Run:
voice
Step 3 Run:
sipservicedata sipaguser-name telephone-number
Step 4 Configure the following functions or services in the SIP service data profile.
Dial tone type dial-tone { normal | special | Normal dial tone type
mwi }
----End
Procedure
l Run the display voice sipaguser [ sipaguser-name ] command to check the configuration
of the SIP AG user.
l Run the display voice sipagusergroup sipag-interface-id [ usergroup-id ] command to
check information about user group.
----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.
IMS
Network
Eth1/0/0 SIPAG
port 2/0/0 port 2/0/2
port 2/0/1
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.
Pre-configuration Tasks
Before configuring the BEST service, complete the following tasks:
l Configuring the router to work in SIP AG mode
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.
l Configuring SIP AG users
NOTE
The SIP AG bound to the BEST service must have user phone numbers configured.
Procedure
1. Run:
system-view
The BEST service takes effect only after you save the configuration and restart the router.
8. Configure a call prefix.
NOTE
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. Configure a SIP server in the PBX view.
10. In the voice view, run:
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 Configuring a SIP AG interface and a SIP AG user
Procedure
Step 1 Run:
system-view
Step 2 Run:
voice
Step 3 Run:
sipag sipag-interface-id
Step 4 Run:
reset
CAUTION
Exercise caution when you run this command because resetting a SIP AG interrupts running
services.
----End
Context
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
IMS
Network
GE1/0/0 SIPAG
port 2/0/0 port 1/0/0
port 2/0/1
Configuration Roadmap
The configuration roadmap is as follows:
1. Configure the AR1200 to work in SIP AG mode.
2. Create SIP AG interface and set parameters for the SIP AG interface.
3. Create SIP AG users and set parameters of SIP AG users.
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
l IP address of GigabitEthernet1/0/0: 1.1.1.1/24
l IP address and port number of the primary proxy server: 2.2.2.2/5060
l SIP AG's ports directly connected to user terminals: port 2/0/0, port 2/0/1, and port 1/0/0
l 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 AR1200 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
<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
Run the display voice sipag [ sipag-interface-id { running | config } ] command to view detailed
information about the SIP AG.
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
#
#
interface GigabitEthernet1/0/0
ip address 1.1.1.1 255.255.255.0
#
return
Networking Requirements
As shown in Figure 1-4, the Router functions as the SIP AG and uses a PRA trunk to connect
to user terminals connected to the PBX to transmit voice, data, and multimedia services. The
PBX can be an IP PBX (the AR router can work in PBX mode) or a digital stored program
control switch.
NOTE
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:
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
l IP address of GigabitEthernet1/0/0: 1.1.1.1/24
l IP address and port number of the primary proxy server: 2.2.2.2/5060
l Phone number of POTSA: 11111111
l Phone number of POTSB: 11112222
Procedure
Step 1 Configure the AR1200 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
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 : 1
Dynamic signalling IP address name :
Signalling IP : 1.1.1.1
Signalling port : 5060
Dynamic media IP address name :
Media IP : 1.1.1.1
Transfer mode : UDP
Primary proxy IP 1 : 2.2.2.2
Primary proxy IP 2 : 255.255.255.255
Secondary proxy IP 1 : 255.255.255.255
Secondary proxy IP 2 : 255.255.255.255
Primary proxy port : 5060
Secondary proxy port : 65535
Primary proxy domain name :
Secondary proxy domain name :
Proxy address mode : IP
Home domain name : huawei.com
SIP profile index : 1: Default
Service logic index : 0: Default
Server Address DHCP option : 0: None
Description :
AG domain name :
Phone context :
Register URI :
Conference factory URI :
Subscribe to UA profile : Enable
Subscribe to reg state : Disable
Subscribe to MWI : Disable
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
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.
IMS
Network
GE1/0/0
SIPAG
port 2/0/0 port 2/0/2
Router
POTS POTS
1000 1001
Configuration Roadmap
The configuration roadmap is as follows:
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
l IP address of Ethernet1/0/0: 10.10.1.171/24
l IP address and port number of the primary proxy server for the SIP AG: 10.10.1.205, 5066
l SIP AG's ports directly connected to user terminals: port2/0/0 and port2/0/1
l Phone numbers of user terminals directly connected to port2/0/0 and port2/0/1 of the SIP
AG: 1000 and 1001
l Media IP address of the SIP server: 192.168.1.3/24
Procedure
Step 1 Configure the AR1200 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...
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
base-telno 1000
agid 1
#
return
2 PBX Configuration
This chapter describes the concepts and configuration of the Private Branch Exchange (PBX),
and provides configuration examples.
2.1 PBX Overview
This section describes the concept of PBX and PBX-related terms.
2.2 PBX Features Supported by the AR1200
The AR1200 provides the following PBX features: SIP server, CDR server, trunk/trunk group,
call routing, individual services, and group services.
2.3 PBX Configuration Procedures
This section provides PBX configuration procedures.
2.4 (Optional) Setting Parameters for Voice Interfaces
This section describes how to set parameters for voice interfaces.
2.5 Setting PBX Parameters
This section describes how to set PBX parameters.
2.6 Configuring an Enterprise
After an enterprise is created, you can configure the Centrex group, prefix, and DN set.
2.7 Configuring a Call Prefix
A call prefix is located at the beginning of a called number.
2.8 Configuring a PBX User
A user identifier includes the user number and DN set.
2.9 (Optional) Configuring a SIP Server
A SIP server is the main component of the IP PBX and establishes SIP sessions.
2.10 (Optional)Configuring SBC Proxy Function
This section describes how to configure the SBC proxy function to implement voice services
for SIPUEs.
2.11 (Optional)Configuring a CDR Server
The call detail record (CDR) of users can be queried in real time, and the CDR data can be
analyzed by using a third-party tool so that users can quickly know call fees.
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 AR1200 can function as a PBX to provide traditional PBX functions and IP PBX functions.
NOTE
l Among the AR1200 series routers, only the AR1220Vs and AR1220VWs support the voice features.
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.
l Centrex and enterprise
A central office exchange service (Centrex) group is a group of users on a PBX or users
connected to PBXs in different areas. The same service is provided for users in the same
Centrex group are provided the same service.
Each user in a Centrex group has two numbers, a private number and a long number. Users
in a Centrex group use private numbers to call each other by default. A user in the Centrex
group must dial the call prefix when calling a user out of the Centrex group. When
configuring or modifying attributes of Centrex users, you must specify the Centrex number.
An enterprise has one or more Centrex groups. Each Centrex group belongs to only one
enterprise. Users in an enterprise may belong to a Centrex group or not belong to any
Centrex group.
Centrex groups and enterprises identify users at the service layer, whereas DN sets identify
users at the switching layer.
l 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
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 Trunk and trunk group
A trunk is a logical link between two exchange offices. Inter-office calls must be transmitted
by trunks. The AR1200 supports the following trunks: AT0, PRA, SIP, H323, and E1R2.
A trunk group is a set of trunks for inter-office calls with same attributes.
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.
l 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.
l 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.
l 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.
l PSTN user
PSTN users are Plain Old Telephone Service (POTS) users.
l 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 AR1200, configure a SIP server
on the AR1200.
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 AR1200.
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 AR1200.
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
Reject anonymous call Allows a user to reject a call when the calling
number is not displayed and plays a voice
prompt to the calling user.
Personal ring back tone (RBT) Uses user-defined music or sound effects to
replace the common ringback tone, and plays
the music or sound effects to calling users.
Group services
Service Type Description
PBX line selection When a calling party dials the access number
of a PBX group, the system uses the
configured line selection mode to connect the
call to a user in the group.
One number link you (ONLY) When the calling party calls ONLY number
of the called party, multiple terminals of the
called party ring according to the configured
rules, and the called party can select one
terminal to answer the incoming call.
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:
CM&BEST
IVR (Interactive
License or CT This license implements the IVR navigation
Voice Response)
(Call Trunk) function.
License
license
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.
FXS1/0/0 FXS1/0/1
User A User B
SIPUE
Figure 2-2 shows the configuration flowchart for intra-office calls. After mandatory
configurations are complete, users can call each other.
Start
End
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.
Start
End
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 AR1200, including physical
attributes, electrical attributes, and KC attributes.
Pre-configuration Tasks
Before setting parameters for an FXS interface, complete the following task:
Data Preparation
To set parameters for an FXS interface, you need the following data.
No. Data
1 Polarity reversal pulse level width, polarity reversal mode, and dialing mode
3 High-level pulse width, low-level pulse width, KC accounting mode, and voltage
operating
Procedure
Step 1 Run:
system-view
Step 2 Run:
voice
Step 3 Run:
port fxs slotid/subcardid/portid
kc voltage voltage
----End
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 AR1200, including the gain,
impedance, ring current, and feed.
Pre-configuration Tasks
Before setting parameters for an FXO interface, complete the following task:
l Configuring the AR1200 to work in PBX mode
l Ensuring that the voice 4FXS1FXO board is working properly
Data Preparation
To set parameters for an FXO interface, you need the following data.
No. Data
Procedure
Step 1 Run:
system-view
NOTE
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 }
----End
Applicable Environment
A basic rate access (BRA) interface connects to an ISDN phone. On the AR1200, 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 Configuring the AR1200 to work in PBX mode
l Ensuring that the 2BST board is working properly
Data Preparation
To set parameters for a BRA interface, you need the following data.
No. Data
1 Interface working mode, automatic deactivation delay, and Layer 1 activation mode
Procedure
Step 1 Run:
system-view
Step 2 Run:
voice
Step 3 Run:
port bra slotid/subcardid/portid
Step 4 Run:
l2-monitor enable
Step 5 Run:
work-mode { p2p | p2mp }
Step 6 Run:
remote-power enable
Step 7 Run:
auto-deactive enable
Step 8 Run:
auto-deactive delay delay
Step 9 Run:
active-mode { unstable | stable }
Step 10 Run:
alarm enable
----End
Applicable Environment
A PRA interface connects to a PBX or PSTN network. On the AR1200, 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:
Data Preparation
To set parameters for a PRA interface, you need the following data.
No. Data
Procedure
Step 1 Run:
system-view
Step 6 Run:
crc4 enable
Step 7 Run:
crc-alarm-threshold { es es-threshold | cses cses-threshold | dm dm- threshold }
Step 8 Run:
l2-monitor enable
Step 9 Run:
pcm-alarm
Step 10 Run:
signal { CCS | CAS }
----End
Applicable Environment
You can configure the AR1200 to work in SIP AG or PBX mode. Before configuring PBX
service features, configure the AR1200 to work in PBX mode. You can run the display voice
service-mode command to view the working mode of the AR1200. If the AR1200 works in SIP
AG mode, delete the SIP AG configurations and configure the AR1200 to work in PBX mode.
If the AR1200 works in PBX mode, skip this configuration.
Pre-configuration Tasks
Before configuring the AR1200 to work in PBX mode, complete the following task:
l Configuring IP addresses and routing protocols for interfaces to ensure connectivity
Procedure
Step 1 Run:
system-view
After the AR1200 is configured to work in PBX mode, restart the AR1200 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
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
Step 2 Run:
voice
Step 3 Run:
voip-address media interface interface-type interface-number { ip-address |
dynamic }
Step 4 Run:
voip-address signalling interface interface-type interface-number { ip-address |
dynamic }
----End
Applicable Environment
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 PBX parameters, complete the following task:
l Configuring IP addresses and routing protocols for interfaces to ensure connectivity
Data Preparation
To set PBX parameters, you need the following data.
No. Data
1 Country/Region identifier
No. Data
2 Country code
5 MWI mode
7 AC amplitude of the ringing current, frequency of the ringing current, and cadence
ratio
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
Step 2 Run:
voice
Step 3 Run:
country { brazil | britain-msfuk | britain-etsi | bulgaria | china-hongkong |
china-mainland | egypt | france | singapore | thailand }
----End
Context
Before configuring a country/region code, run the display voice country-code command to
view the pre-configured country/region codes on the AR1200.
Procedure
Step 1 Run:
system-view
Step 2 Run:
voice
Step 3 Run:
country-code country-code-value [ country-name country-name | international-prefix
international-prefix-value | national-prefix national-prefix-value ]*
NOTE
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-prefix-
value.
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 }*
----End
Context
Before configuring the area code of toll calls, run the display voice country-code country-code-
value command to view the pre-configured area codes.
Procedure
Step 1 Run:
system-view
Step 2 Run:
voice
Step 3 Run:
area-code country-code-value area-code-value [ area-name area-name-value ]
----End
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.
l 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.
Procedure
Step 1 Run:
system-view
----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
----End
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
----End
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 AR1200 to meet local standards.
Procedure
Step 1 Run:
system-view
----End
Procedure
Step 1 Run:
system-view
----End
Context
When the CLIP service is registered, CLIP parameters in offhook state need to be configured
on the AR1200 so that the AR1200 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
Step 2 Run:
voice
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
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 AR1200 so that the AR1200 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
Step 2 Run:
voice
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
The number of bits of the FSK synchronization mask in onhook state is set.
----End
Context
The AR1200 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
media-bandwidth-control enable
The AR1200 is enabled to restrict calls when the uplink bandwidth is enabled.
l Run:
media-bandwidth-control maximum max-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.
l Run the display voice user-defined-ring [ring-index ] command to check user-defined
ring information.
l Run the display voice clip command to check CLIP parameters.
----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 Configuring the AR1200 to work in SIP AG mode
Data Preparation
To set SIP protocol stack parameters, you need the following data.
No. Data
Procedure
Step 1 Run:
system-view
Step 2 Run:
voice
Step 3 Run:
sip
l Run:
t1 t1period
----End
Procedure
Step 1 Run:
system-view
Step 2 Run:
voice
Step 3 Run:
sip
l Run:
field-header user-agent user-agent-head
----End
Procedure
l Run the display voice sip command to check SIP parameters.
----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 Configuring the AR1200 to work in SIP AG mode
l Configuring IP addresses and routing protocols for interfaces to ensure connectivity
Data Preparation
To set DSP parameters, you need the following data.
No. Data
2 Default DSP channel code type and default interval at which the DSP channel
packetizes RTP packets
No. Data
3 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
4 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
5 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
6 Interval at which a DSP channel sends RTCP packets and threshold for the number
of severe degrade seconds
8 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
----End
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
Step 2 Run:
voice
Step 3 Run:
dsp-attribute
Step 4 Run:
plc enable
----End
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
Step 2 Run:
voice
Step 3 Run:
dsp-attribute
Step 4 Run:
silence enable
----End
Procedure
Step 1 Run:
system-view
Step 2 Run:
voice
Step 3 Run:
dsp-attribute
Step 4 Run:
autovbd { auto | host-controlled }
NOTE
By default, the VBD switching mode is host-controlled.
----End
Procedure
Step 1 Run:
system-view
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
Step 2 Run:
voice
Step 3 Run:
dsp-attribute
----End
Procedure
Step 1 Run:
system-view
Step 2 Run:
voice
Step 3 Run:
dsp-attribute
----End
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
Step 2 Run:
voice
Step 3 Run:
dsp-attribute
The RTP Control Protocol Extended Reports (RTCP XR) function is enabled.
l Run:
rtcp sev-degradethreshold sev-degradethresholdval
----End
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
Step 2 Run:
voice
----End
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
data-event data-event-value
----End
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.
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
----End
Procedure
l Run the display voice dsp-attribute command to check the DSP configuration.
l Run the display voice dsp state { slot/dsp-index | channel slot/dsp-index/channel }
command to check the status of a DSP or DSP channel.
l 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.
Pre-configuration Tasks
Before configuring an enterprise, complete the following task:
l Configuring the AR1200 to work in PBX mode
Data Preparation
To configure an enterprise, you need the following data.
No. Data
1 Enterprise name
3 DN set
Procedure
Step 1 Run:
system-view
The following is the default maximum number of Centrex groups that can be configured in the
enterprise view for different devices:
l AR1220: 4
l AR1240: 8
Step 6 Run:
dn-set dn-set-name [ description description ]
A DN set is configured.
----End
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
1 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
Step 2 Run:
voice
Step 3 Run:
callprefix callprefix-name
Step 4 Run:
prefix prefix
Step 5 Run:
enterprise enterprise-name [ dn-set dn-set-name ] [ centrex centrex-name ]
Step 6 Run:
call-type category callcategory attribute attribute
The call type and call attribute are configured for the call prefix.
Step 7 Run:
digit-length maximum-length-value minimum-length-value
NOTE
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.
----End
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 the device to work in PBX mode
l Setting global PBX parameters
l Configuring an enterprise
Data Preparation
To configure an enterprise, you need the following data.
No. Data
1 User name
Procedure
Step 1 Run:
system-view
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.
By default, no country code, area code, or telephone number is configured for a PBX user.
Step 6 Run:
dn-set dn-set-name
----End
Usage Scenario
If SIP users need to communicate with each other through the AR1200, configure a SIP server.
Pre-configuration Tasks
Before configuring a SIP server, complete the following tasks:
Data Preparation
To configure a SIP server, you need the following data.
No. Data
Procedure
Step 1 Run:
system-view
Step 2 Run:
voice
Step 3 Run:
sipserver
Step 4 Run:
signaling-address{ ip ip-address | addr-name addr-name-value } port port-value
A signaling IP address and signaling port number are configured for the SIP server.
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.
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 }
Step 8 Run:
register-uri uri
Step 9 Run:
home-domain domain
Step 10 Run:
reset
CAUTION
Exercise caution when you run this command because resetting a SIP server interrupts running
services.
----End
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 trunk
SIP UE Router D
(Travelling staff)
Router A
Headquarters
User A
Enterprise A
Pre-configuration Tasks
Before configuring the SBC proxy function, complete the following tasks:
Data Preparation
To configure the SBC proxy function, you need the following data.
No. Data
1 Media trunk type, and signaling and media proxy policies for the SIP server
Procedure
Step 1 Run:
system-view
Step 2 Run:
voice
Step 3 Run:
sbc media-relay interface interface-type interface-number relay-type
Step 4 Run:
sipserver
Step 5 Run:
sbc signalling-proxy value
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.
Step 6 Run:
sbc media-proxy value
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.
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
Router A ISDN
IP network
SIP UE
(Travelling staff)
Pre-configuration Tasks
Before configuring the SBC proxy function, complete the following tasks:
Data Preparation
To configure the SBC proxy function, you need the following data.
No. Data
1 Media trunk type, and signaling and media proxy policies for the SIP server
Procedure
Step 1 Run:
system-view
NOTE
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
NOTE
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
----End
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.
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
Router A Headquarters
SIP trunk
Enterpeise A
Pre-configuration Tasks
Before configuring the SBC proxy function, complete the following tasks:
l Configuring the SIP server
l Configuring the SIP trunk group
l Configuring SIP UEs
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.
No. Data
1 Media proxy port number, media trunk type, and signaling and media proxy policies
for the SIP server
Procedure
Step 1 Run:
system-view
Step 4 Configure the SBC proxy function for the SIP server.
l Run:
sipserver
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
l Run:
sbc mapped-media-proxy-port-start port-value
The media proxy port number mapping the SIP server is configured.
l Run:
return
Step 5 Configure the SBC proxy function for the SIP trunk group.
l 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 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
Applicable Environment
To record and display CDRs of users, specify a CDR server on the AR1200.
Pre-configuration Tasks
Before configuring a CDR server, complete the following tasks:
l Configuring the AR1200 to work in PBX mode
l Assigning IP addresses to interfaces
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}
----End
Usage Scenario
As shown in Figure 2-8, the AR1200 functions as a PBX and needs to communicate with other
PBXs or external networks. The AR1200 supports the following connection modes:
l Connects to a PSTN network through an AT0 trunk.
l Connects to a PSTN network through a PRA trunk.
l Connects to an IMS network through a SIP trunk.
l Connects to another PBX through a PRA trunk.
l Connects to another PBX through a E1R2 trunk.
l Connects to another PBX through a SIP trunk.
l Connects to an IMS network through an H323 trunk.
l Connects to another PBX through an H323 trunk.
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.
IMS PSTN
SIP/H323 AT0/PRA
AR
SIP/PRA/E1R2/H323
PBX
Pre-configuration Tasks
Before configuring a trunk group, complete the following tasks:
Data Preparation
To configure a trunk group, you need the following data.
No. Data
1 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 call-
in 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
2 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
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 AR1200 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.
l When the AR1200 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.
l When the AR1200 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.
l When the AR1200 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.
l When the AR1200 connects to the remote network through an AT0 trunk, set the signaling
type to FXO.
l When the AR1200 connects to the remote network through a SIP trunk, set the signaling
type to SIP.
l If the AR1200 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
----End
Procedure
Step 1 Run:
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
----End
Context
When the trunk group type is PRA, perform the following operations.
Procedure
Step 1 Run:
system-view
----End
Context
When the trunk group type is SIP, perform the following operations.
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 secondary-
ip-value secondary-port-value ]
The remote IP address and port are configured for the SIP trunk group.
l Run:
register-uri register-uri-value
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.
l Run:
redundancy-start-mode { ordinary2198 | smart2198 }
The PBX is enabled to transmit VBD using RFC 2198 redundancy transmission.
l Run:
nte-flashhook enable
----End
Context
When an H323 trunk is used, the AR1200 can function as the gatekeeper and gateway and is
used in the following scenarios:
l 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.
N e tw
IP o rk
R o u te rA R o u te rB
U s e rA GW GW U se rB
G W s a re p e e rs
l 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
AR1200 does not function as the gateway, that is, the AR1200 does not send registration
packets.
IP o rk
N e tw
R o u te rB
U se rA GW
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
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
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 AR1200 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 AR1200 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
----End
Context
When the trunk group type is E1R2, perform the following operations:
Procedure
Step 1 Run:
system-view
voice
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
Context
When the SIP trunk registration mode is trunk circuit registration, perform the following
operations.
Procedure
Step 1 Run:
system-view
----End
Procedure
Step 1 Run:
system-view
----End
Procedure
Step 1 Run:
system-view
----End
Procedure
Step 1 Run:
system-view
Step 2 Run:
voice
Step 3 Run:
trunk-group name
Step 4 Run:
trunk-e1r2 slotid/subcardid/portid
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 Configuring the SIP trunk group
Procedure
Step 1 Run:
system-view
Step 2 Run:
voice
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 AR1200, which meet the following requirements:
l Calls between users in the headquarters and branch are connected through the SIP trunk.
l 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.
Branch
User B
PSTN
AR
Enterprise A Carrier A
SIP
PSTN
Carrier B
IP
AR
Headquarters
PBX
Enterprise A
User A
Pre-configuration Tasks
Before configuring a call route, complete the following tasks:
l Configuring a Call Prefix
l Configuring a trunk group
Data Preparation
To configure a call route, you need the following data.
No. Data
2 Enterprise, Centrex group, and DN set to which the calling number is bound, and
calling condition
No. Data
Procedure
Step 1 Run:
system-view
----End
Applicable Environment
After the abbreviated dialing service is configured on the AR1200, 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.
Pre-configuration Tasks
Before configuring the abbreviated dialing service, complete the following tasks:
l Configuring the AR1200 to work in PBX mode
l Configuring a PBX user
Data Preparation
To configure the abbreviated dialing service, you need the following data.
No. Data
1 Abbreviated code
Procedure
Step 1 Run:
system-view
----End
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:
Data Preparation
To configure the call-out restriction service, you need the following data.
No. Data
1 User name
Procedure
Step 1 Run:
system-view
Step 2 Run:
voice
Step 3 Run:
pbxuser name
Step 4 Run:
service-right cba enable
Step 5 Run:
service cba { all | idd_cdd | idd }
----End
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.
l 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.
l Call Forwarding Offline (CFO): Incoming calls of a CFO user are forwarded to a preset
number when the user is offline.
l 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:
Data Preparation
To configure the call forwarding service, you need the following data.
No. Data
2 Forwarded-to number
Procedure
Step 1 Run:
system-view
----End
Applicable Environment
To prevent a user from dialing a number, configure the number barring service.
Pre-configuration Tasks
Before configuring the number barring service, complete the following tasks:
Data Preparation
To configure the number barring service, you need the following data.
No. Data
2 Restricted number
Procedure
Step 1 Run:
system-view
Step 2 Run:
voice
Step 3 Run:
pbxuser name
Step 4 Run:
service-right dlc enable
Step 5 Run:
service dlc dlc-telno
----End
Applicable Environment
The DND service allows a user to reject all incoming calls during a certain period.
Pre-configuration Tasks
Before configuring the DND service, complete the following tasks:
Data Preparation
To configure the DND function, you need the following data.
No. Data
2 DND tone
Procedure
Step 1 Run:
system-view
Step 2 Run:
voice
Step 3 Run:
pbxuser name
Step 4 Run:
service-right dnd enable
Step 5 Run:
service dnd [ tone-id tone-id ]
----End
Applicable Environment
To prevent anonymous calls, configure the RAC service.
Pre-configuration Tasks
Before configuring the RAC service, complete the following tasks:
Data Preparation
To configure the RAC service, you need the following data.
No. Data
2 RAC tone
Procedure
Step 1 Run:
system-view
Step 2 Run:
voice
Step 3 Run:
pbxuser name
Step 4 Run:
service-right rac enable
Step 5 Run:
service rac [ tone-id tone-id ]
----End
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:
Data Preparation
To configure the remote office service, you need the following data.
No. Data
Procedure
Step 1 Run:
system-view
Step 4 Run:
service-right remote-office enable
Step 5 Run:
service remote-office telno { centrex centrex-name short-num | [ country-code
countrycode ] [ area-code areacode ] telno }
By default, the destination number of the remote office service is not configured.
----End
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:
Data Preparation
To configure the secretary service, you need the following data.
No. Data
Procedure
Step 1 Run:
system-view
Step 2 Run:
voice
Step 3 Run:
pbxuser name
Step 4 Run:
service-right scr enable
Step 5 Run:
service scr telno { centrex centrex-name short-num | countrycode areacode telno }
----End
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:
Data Preparation
To configure the wake-up service, you need the following data.
No. Data
1 User name
2 Wake-up time
Procedure
Step 1 Run:
system-view
----End
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 Configuring the AR1200 to work in PBX mode
l Configuring a PBX user
Data Preparation
To configure the RBT service, you need the following data.
No. Data
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
Step 2 Run:
voice
Step 3 Run:
pbxuser name
Step 4 Run:
service-right crbt enable
Step 5 Run:
service crbt [ condition { caller-trunkgroup trunkgroup-name | caller-telno caller-
telno-value | time-period { from from-date [ from-time ] | to to-date [ to-time ] }
* | time-repeat { { yearly | monthly }begin-date [ begin-time ] [ end-date [ end-
----End
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 Configuring the AR1200 to work in PBX mode
l Configuring a PBX user
Data Preparation
To configure the SCR service, you need the following data.
No. Data
2 Rejected number
Procedure
Step 1 Run:
system-view
----End
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 Configuring a PBX user
Data Preparation
To configure the SCA service, you need the following data.
No. Data
Procedure
Step 1 Run:
system-view
Step 3 Run:
pbxuser name
----End
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 AR1200.
Pre-configuration Tasks
Before configuring the call interception service, complete the following tasks:
Data Preparation
To configure the call interception service, you need the following data.
No. Data
1 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
----End
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 Configuring the AR1200 to work in PBX mode
l Creating an enterprise
l Creating a Centrex group
Data Preparation
To the distinctive ringing service, you need the following data.
No. Data
1 Distinctive ringing service plan name, and enterprise and Centrex group bound to the
plan
2 (Optional) Ring IDs for intra-group calls, local calls, national toll calls, and
international toll calls
Procedure
Step 1 Run:
system-view
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 ]
----End
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 Configuring the AR1200 to work in PBX mode
l Creating an enterprise
l Creating a trunk group
Data Preparation
To configure the enterprise RBT service, you need the following data.
No. Data
1 Name of the RBT service plan and enterprise bound to the plan
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
Step 2 Run:
voice
Step 3 Run:
enterprise enterprise-name
Step 4 Run:
service crbt [ condition { caller-telno caller-telno-value | caller-trunkgroup
trunkgroup-name | time-period { from from-date [ from-time ] | to to-date [ to-
time ] } * | time-repeat { yearly begin-date [ begin-time ] [ end-date [ end-
time] ] | monthly begin-date [ begin-time ] [ end-date [ end-time] ] | weekly begin-
weekday [ begin-time ] [ end-weekday [ end-time ] ] | daily begin-time [ end-
time] } } ] file-name file-name-value
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
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.
No. Data
2 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
3 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
No. Data
4 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
5 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
Procedure
Step 1 Run:
system-view
Step 2 Run:
voice
Step 3 Run:
ivr-file file-name-value
Step 4 Run:
enterprise enterprise-name
----End
Procedure
Step 1 Run:
system-view
NOTE
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
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
Procedure
Step 1 Run:
system-view
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
Procedure
Step 1 Run:
system-view
The calling party, validity period, and time repeat mode are configured for the IVR group.
Step 7 Run:
console-telno value
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 maximum-queue-value | queue-time queue-time-
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 }
The flag indicating whether the IVR group can connect to the switchboard is set.
l Run:
navigate-menu menu-name
----End
Procedure
Step 1 Run:
system-view
An operation code input when the IVR navigation group takes the menu action is configured.
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 }
----End
Procedure
l Run the display voice ivr-action [ ivr-action-name ] command to view information about
one IVR action or all IVR actions.
l Run the display voice ivr-file [ ivr-file-name ] command to view information about one
IVR file or all IVR files.
l Run the display voice ivr-menu [ ivr-menu-name ] command to view information about
one IVR menu or all IVR menus.
l Run the display voice pbxusergroup [ pbxusergroup-name ] command to check the
configuration of the PBX user group.
----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:
Data Preparation
To configure the number change service, you need the following data.
No. Data
2 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
----End
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:
Data Preparation
To configure the pre-routing number change service, you need the following data.
No. Data
2 Enterprise, DN set, Centrex group, and call prefix bound to the pre-routing number
change plan
3 Calling numbers that need to be changed before routing, calling number change rule,
and called number change rule
4 (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 }
----End
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:
Data Preparation
To configure the post-routing number change service, you need the following data.
No. Data
Procedure
Step 1 Run:
system-view
Step 2 Run:
voice
Step 3 Run:
afterroute-change name
Step 4 Run:
callprefix callprefix-name
Step 5 Run:
condition caller-telno { [ country-code country-code-value ] [ area-code area-code-
value ] telno-value | centrex centrex-telno-value }
Step 6 Run:
trunk-group trunk-group-name
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 }
----End
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 Configuring the AR1200 to work in PBX mode
l Creating an enterprise
l Creating a trunk group
Data Preparation
To configure the PBX line selection service, you need to the following data.
No. Data
Procedure
Step 1 Run:
system-view
Step 2 Run:
voice
Step 3 Run:
pbxusergroup user-group-name hunt [ enterprise enterprise-name ]
Step 4 Run:
dn-set dn-set-value
Step 5 Run:
access-telno { centrex centrex-name centrex-telno-value | [ country-code country-
code-value ] [ area-code area-code-value ] telno-value }
NOTE
The access number must be a local user number.
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:
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
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:
Data Preparation
To configure simultaneous ringing, you need the following data.
No. Data
Procedure
Step 1 Run:
system-view
Step 2 Run:
voice
Step 3 Run:
pbxusergroup user-group-name ring-parallel [ enterprise enterprise-name ]
Step 4 Run:
dn-set dn-set-value
Step 5 Run:
access-telno { centrex centrex-name centrex-telno-value | [ country-code country-
code-value ] [ area-code area-code-value ] telno-value }
----End
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 Configuring the AR1200 to work in PBX mode
l Creating an enterprise
Data Preparation
To configure sequential ringing, you need the following data.
No. Data
4 (Optional) Ringing duration and line selection mode of the sequential ringing group
Procedure
Step 1 Run:
system-view
Step 4 Run:
dn-set dn-set-value
Step 5 Run:
access-telno { centrex centrex-name centrex-telno-value | [ country-code country-
code-value ] [ area-code area-code-value ] telno-value }
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:
Data Preparation
To configure the ONLY service, you need the following data.
No. Data
No. Data
2 Enterprise and DN set bound to the ONLY service and access number of the ONLY
service
3 (Optional) Ringing duration, ringing mode, and line selection mode of the ONLY
service
Procedure
Step 1 Run:
system-view
Step 2 Run:
voice
Step 3 Run:
pbxusergroup user-group-name unicall [ enterprise enterprise-name ]
Step 4 Run:
dn-set dn-set-value
Step 5 Run:
access-telno { centrex centrex-name centrex-telno-value | [ country-code country-
code-value ] [ area-code area-code-value ] telno-value }
The ringing duration, ringing mode, and line selection mode are set for the ONLY service.
----End
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 Configuring the trunk group
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
Applicable Environment
The DISA service can be used in the following scenarios:
l Controlling incoming calls made by external users
To prevent unauthorized users to connect to an enterprise's PBX, the DISA service can be
configured on the PBX to enable the PBX to authenticate PBX users. To use the voice
service provided by the PBX, an external user needs to dial a specified phone number of
the enterprise and then enter authentication information as prompted. The user can use
voice services provided by the PBX only after being authenticated by the PBX. As shown
in Figure 2-12, the PBX has the DISA service configured, with the access number 800.
When user A dials 25600000, a voice message is played, prompting user A to enter
authentication information. After user A is authenticated, the PBX provides voice services
to user A.
NOTE
PSTN
User A
AT0 trunk
External number:
25600000
Port 1/0/4 Internal number: 800
PBX
User B User C
PSTN
User A
User B User C
Pre-configuration Tasks
Before configuring the DISA service, complete the following task:
Data Preparation
To configure the DISA service, you need the following data.
No. Data
2 Enterprise name
Procedure
Step 1 Create an account set and configure the authentication mode for the account set.
1. Run:
system-view
Step 3 Run:
quit
A DISA service profile is created and the DISA service view is displayed.
2. Run:
dn-set dn-set-name
5. (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
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.
IMS/IP
network
PBX
User A User B
SIPAG
(POTS) (SIPUE)
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 Configuring PBX users
Procedure
l Perform the following steps to configure the intermediate conference service for the POTS
and SIPUE users connected to the PBX:
1. Run:
system-view
The call type and call attribute are configured for the intermediate conference service.
8. Run:
quit
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 three-
party 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.
l 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
Networking Requirements
As shown in Figure 2-15, an enterprise has POTS users: User A, User B, and User C. The
requirements are as follows:
l 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.
l 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."
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. Set the service mode to PBX.
2. Configure the default country code and area code.
3. Configure the enterprise, DN set, and RTB file.
4. Configure prefixes.
5. Configure PBX users.
6. Configure a trunk and a trunk group for inter-office calls.
7. Configure a call route and a post-routing number change plan.
8. Configure an IVR group.
Data Preparation
To complete the configuration, you need the following data:
The country code and region code in China are used as an example.
l Extension numbers of User A, User B, and User C: 800, 801, and 802 (the access number
of the IVR group is 800)
l Enterprise that User A, User B, and User C belongs to: hw
l DN set: local
l Intra-office and inter-office call prefixes: 8 and 9
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...
# Configure User B.
[RouterA-voice] pbxuser 801 pots enterprise hw
[RouterA-voice-pbxuser-801] port 1/0/1
[RouterA-voice-pbxuser-801] telno country-code 86 area-code 25 801
[RouterA-voice-pbxuser-801] dn-set local
[RouterA-voice-pbxuser-801] call-right in international-toll out international-
toll
[RouterA-voice-pbxuser-801] quit
# Configure User C.
[RouterA-voice] pbxuser 802 pots enterprise hw
[RouterA-voice-pbxuser-802] port 1/0/2
[RouterA-voice-pbxuser-802] telno country-code 86 area-code 25 802
[RouterA-voice-pbxuser-802] dn-set local
[RouterA-voice-pbxuser-802] call-right in international-toll out international-
toll
[RouterA-voice-pbxuser-802] quit
----End
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
called del 7 1
#
return
Networking Requirements
As shown in Figure 2-16, enterprise A and enterprise B are located in the same industry park.
User A and User B belong to enterprise A, and User C and User D belong to enterprise B. The
AR logically isolates voice services of the two enterprises. Internal calls of the enterprises are
connected through their respective virtual PBXs on the AR, and outgoing calls from the
enterprises are connected to external users through one port. This networking helps enterprises
lower investment on equipment and reduces the number of access nodes on the carrier network.
The requirements are as follows:
l 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.
l 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 Port 1/0/3
Port Port
User A 1/0/1 User D
1/0/2
User B User C
Enterprise A Enterprise B
Campus network
Configuration Roadmap
The configuration roadmap is as follows:
Data Preparation
To complete the configuration, you need the following data:
The country code and region code in China are used as an example.
l Numbers of User A, User B, User C, and User D: 2000, 2001, 3000, and 3001
l Signaling and media IP addresses: 192.168.1.3
l Enterprise huawei to which user A belongs, DN set local, intra-office call prefix 2, and
inter-office call prefix 8
l Enterprise huawei to which user B belongs, DN set local1, intra-office call prefix 3, and
inter-office call prefix 9
l IP address and port number of the IMS: 192.168.1.1 and 5060
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...
# Configure User B.
[Huawei-voice] pbxuser 2001 pots enterprise hw
[Huawei-voice-pbxuser-2001] port 1/0/1
[Huawei-voice-pbxuser-2001] telno country-code 86 area-code 25 2001
[Huawei-voice-pbxuser-2001] dn-set local
[Huawei-voice-pbxuser-2001] call-right in international-toll out international-
toll
[Huawei-voice-pbxuser-2001] quit
# Configure User C.
[Huawei-voice] pbxuser 3000 pots enterprise hw1
[Huawei-voice-pbxuser-3000] port 1/0/2
[Huawei-voice-pbxuser-3000] telno country-code 86 area-code 25 3000
[Huawei-voice-pbxuser-3000] dn-set local1
[Huawei-voice-pbxuser-3000] call-right in international-toll out international-
toll
[Huawei-voice-pbxuser-3000] service-right call-transfer enable
[Huawei-voice-pbxuser-3000] quit
# Configure User D.
[Huawei-voice] pbxuser 3001 pots enterprise hw1
[Huawei-voice-pbxuser-3001] port 1/0/3
[Huawei-voice-pbxuser-3001] telno country-code 86 area-code 25 3001
[Huawei-voice-pbxuser-3001] dn-set local1
[Huawei-voice-pbxuser-3001] call-right in international-toll out international-
toll
[Huawei-voice-pbxuser-3001] quit
[Huawei-voice-trunkgroup-sipat01] home-domain
huawei.com
[Huawei-voice-trunkgroup-sipat01] register-id 56623001
[Huawei-voice-trunkgroup-sipat01] trunk-sipat0 56623001 default-called-telno
3000
[Huawei-voice-trunkgroup-sipat01] reset
Note: Trunkgroup reset succeeds.
[Huawei-voice-trunkgroup-sipat01] quit
----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
#
dsp-
attribute
#
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
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
#
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
Networking Requirements
As shown in Figure 2-17, the headquarters and branch of enterprise A are located in different
areas. RouterA and RouterB function as gateways and are connected through the H323 trunk.
After voice services are deployed on RouterA and RouterB, enterprise users can use the voice
services across areas. Internal users use the AT0 trunk to call external users.
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.
l 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.
Figure 2-17 Networking diagram for configuring calls between the headquarters and branch
Branch Enterprise A
User C User D
Port1/0/0 Port1/0/1
Router B
Port1/0/4 Eth2/0/0
IP network
PSTN
H323 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. Set the service mode to PBX.
2. Configure signaling and media IP addresses.
3. Configure a country code and a region code.
4. Configure the enterprise and DN set.
5. Set H323 system parameters.
6. Configure prefixes.
7. Configure PBX users.
8. Configure trunks, trunk groups, and routes for inter-office calls.
9. Configure a call route and post-routing number change.
Data Preparation
To complete the configuration, you need the following data:
l 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.
l Number of user A: 22223000
l Number of user B: 22223001
l Number of user C: 33333000
l Number of user D: 33333001
l Media and signaling IP address of the headquarters: 192.168.1.1
l Media and signaling IP address of the branch: 192.168.1.2
l Enterprise hw to which user A and user B belong, DN set local, call prefix 2222, inter-
office prefix 9 of the AT0 trunk, and inter-office prefix 20000 between the headquarters
and branch
l Enterprise hw to which user C and user D belong, DN set local, call prefix 3333, inter-
office 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.
# Configure RouterA.
[RouterA] voice
[RouterA-voice] pbx default-country-code 86 default-area-code 25
# Configure RouterB.
[RouterB] voice
[RouterB-voice] pbx default-country-code 86 default-area-code 755
# Configure RouterA.
[RouterA-voice] enterprise hw
[RouterA-voice-enterprise-hw] dn-set local
[RouterA-voice-enterprise-hw] quit
# Configure RouterB.
[RouterB-voice] enterprise hw
[RouterB-voice-enterprise-hw] dn-set local
[RouterB-voice-enterprise-hw] quit
[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
# Configure user B.
[RouterA-voice] pbxuser 22223001 pots enterprise hw
[RouterA-voice-pbxuser-22223001] port 1/0/1
# Configure user C.
[RouterB-voice] pbxuser 33333000 pots enterprise hw
[RouterB-voice-pbxuser-33333000] port 1/0/0
[RouterB-voice-pbxuser-33333000] telno country-code 86 area-code 755 33333000
[RouterB-voice-pbxuser-33333000] dn-set local
[RouterB-voice-pbxuser-33333000] call-right in international-toll out
international-toll
[RouterB-voice-pbxuser-33333000] service-right call-transfer enable
[RouterB-voice-pbxuser-33333000] quit
# Configure user D.
[RouterB-voice] pbxuser 33333001 pots enterprise hw
[RouterB-voice-pbxuser-33333001] port 1/0/1
[RouterB-voice-pbxuser-33333001] telno country-code 86 area-code 755 33333001
[RouterB-voice-pbxuser-33333001] dn-set local
[RouterB-voice-pbxuser-33333001] call-right in international-toll out
international-toll
[RouterB-voice-pbxuser-33333000] quit
----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-
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
trunk-group h323
caller no-change
called del 7 5
#
return
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
Networking Requirements
As shown in Figure 2-18, the headquarters and branch of enterprise A are located in different
areas. RouterA functions as the gateway, and RouterB is a non-Huawei device and functions as
the gatekeeper. RouterA and RouterB are connected through the H323 trunk. After voice services
are deployed on RouterA and RouterB, enterprise users can use the voice services across areas.
Figure 2-18 Networking diagram for configuring calls between the headquarters and branch
Branch Enterprise A
User C User D
Port1/0/0 Port1/0/1
Router B
IP network
H323 trunk
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. Set the service mode to PBX.
2. Configure signaling and media IP addresses.
3. Configure a country code and a region code.
4. Configure the enterprise and DN set.
5. Set H323 attribute.
6. Configure prefixes.
7. Configure PBX users.
8. Configure trunk groups and routes for inter-office calls.
9. Configure a call route and post-routing number change.
Data Preparation
To complete the configuration, you need the following data (data of Router A):
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.
l Number of user A: 22223000 and number of user B: 22223001 Number of user C: 33333000
and number of user D: 33333001
l 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
l 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.
# Configure RouterA.
[RouterA] voice
[RouterA-voice] pbx default-country-code 86 default-area-code 25
# Configure RouterA.
[RouterA-voice] enterprise hw
[RouterA-voice-enterprise-hw] dn-set local
[RouterA-voice-enterprise-hw] quit
# Configure user A.
[RouterA-voice] pbxuser 22223000 pots enterprise hw
[RouterA-voice-pbxuser-22223000] port 1/0/0
[RouterA-voice-pbxuser-22223000] telno country-code 86 area-code 25 22223000
[RouterA-voice-pbxuser-22223000] dn-set local
[RouterA-voice-pbxuser-22223000] call-right in international-toll out
international-toll
[RouterA-voice-pbxuser-22223000] service-right call-transfer enable
[RouterA-voice-pbxuser-22223000] quit
# Configure user B.
[RouterA-voice] pbxuser 22223001 pots enterprise hw
[RouterA-voice-pbxuser-22223001] port 1/0/1
[RouterA-voice-pbxuser-22223001] telno country-code 86 area-code 25 22223001
[RouterA-voice-pbxuser-22223001] dn-set local
[RouterA-voice-pbxuser-22223001] call-right in international-toll out
international-toll
[RouterA-voice-pbxuser-22223001] 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
----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
#
h323-
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 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
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.
l 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.
Figure 2-19 Networking for configuring communication between the headquarters and branch
Branch Enterprise A
User C User D
Port1/0/0 Port1/0/1
Router B
Port1/0/4 Eth2/0/0
PSTN
SIP trunk
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. Set the service mode to PBX.
2. Configure signaling and media IP addresses.
3. Configure the default country code and area code.
4. Configure the enterprise and DN set.
5. Configure the SIP server.
6. Configure prefixes.
7. Configure PBX users.
8. Configure trunks, trunk groups, and routes for inter-office calls.
9. Configure call routes and post-routing number change plans.
Data Preparation
To complete the configuration, you need the following data:
l 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.
l Internal numbers of User A, User B, User C, and User D: 22223000, 22223001, 33333000,
and 33333001
l Signaling and media IP addresses and signaling port number of the headquarters:
192.168.1.1 and 5070
l Signaling and media IP addresses and signaling port number of the branch: 192.168.1.2
and 5070
l Enterprise huawei to which user A and user B belong, DN set local, call prefix 2222, inter-
office call prefix 9 of the AT0 trunk, and inter-office call prefix 20000 between the
headquarters and branch
l Enterprise huawei to which user C and user D belong, DN set local, call prefix 3333, inter-
office 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.
# Configure RouterB.
[RouterB-voice] pbx
[RouterB-voice] enterprise hw
[RouterB-voice-enterprise-hw] dn-set local
[RouterB-voice-enterprise-hw] quit
[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
# Configure User A.
[RouterA-voice] pbxuser 22223000 pots enterprise hw
[RouterA-voice-pbxuser-22223000] port 1/0/0
[RouterA-voice-pbxuser-22223000] telno country-code 86 area-code 25 22223000
[RouterA-voice-pbxuser-22223000] dn-set local
[RouterA-voice-pbxuser-22223000] call-right in international-toll out
international-toll
[RouterA-voice-pbxuser-22223000] service-right call-transfer enable
[RouterA-voice-pbxuser-22223000] quit
# Configure User B.
[RouterA-voice] pbxuser 22223001 pots enterprise hw
[RouterA-voice-pbxuser-22223001] port 1/0/1
[RouterA-voice-pbxuser-22223001] telno country-code 86 area-code 25 22223001
[RouterA-voice-pbxuser-22223001] dn-set local
[RouterA-voice-pbxuser-22223001] call-right in international-toll out
international-toll
[RouterA-voice-pbxuser-22223001] quit
# Configure User C.
[RouterB-voice] pbxuser 33333000 pots enterprise hw
[RouterB-voice-pbxuser-33333000] port 1/0/0
[RouterB-voice-pbxuser-33333000] telno country-code 86 area-code 755 33333000
[RouterB-voice-pbxuser-33333000] dn-set local
[RouterB-voice-pbxuser-33333000] call-right in international-toll out
international-toll
[RouterB-voice-pbxuser-33333000] service-right call-transfer enable
[RouterB-voice-pbxuser-22223001] quit
# Configure User D.
[RouterB-voice] pbxuser 33333001 pots enterprise hw
[RouterB-voice-pbxuser-33333001] port 1/0/1
[RouterB-voice-pbxuser-33333001] telno country-code 86 area-code 755 33333001
[RouterB-voice-pbxuser-33333001] dn-set local
[RouterB-voice-pbxuser-33333001] call-right in international-toll out
international-toll
[RouterB-voice-pbxuser-22223001] quit
----End
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
#
dsp-
attribute
#
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
#
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
#
#
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