NEXO 450/8100/16300

IP PBX

IP PBX Administrator Manual

Version 1.2

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 Table of Contents

1. Introduction 5
1.1 Overview 5

1.2 Product Features 5

1.3 Product Appearance 6

1.4 Scenario of Application 9

2. Installation Guide 10
2.1 Installation Notice 10

2.2 Installation Procedure 10

2.2.1 Connect Drawing 10

3. WEB Interface Configuration 11

3.1 Access NEXO 8100 unit 11

3.2 Parameters Configuration 12

3.3 System Information 13

3.3.1 System Information 13

3.3.2 Extensions Status 14

3.3.3 Trunk Status 14

3.4 Network Configuration 15

3.4.1 LAN Configuration 15

3.4.2 VLAN Configuration 17

3.4.3 ARP Configuration 19

3.4.4 VPN Configuration 20

3.4.5 DDNS Server 21

3.4.6 Static Route 21

3.4.7 DHCP Server 23

3.5 Trunks 24

3.5.1 Physical Trunks(PSTN and GSM Trunks) 24

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 3.5.2 IP Trunk (Peer to Peer Mode) 28

3.5.3 VoIP Trunk 30

3.6 PBX Basic 34

3.6.1 Extensions 34

3.6.2 Feature Codes 45

3.6.3 Speed dial 49

3.6.4 Outbound Routes 50

3.6.5 Parking Lot 54

3.6.6 Time Groups 55

3.6.7 General Preferences 57

3.7 PBX Inbound Call Control 59

3.7.1 Inbound Routes 59

3.7.2 Blacklist 64

3.7.3 IVR 64

3.7.4 Queue 67

3.7.5 Ring Groups 71

3.7.6 Conferences 73

3.7.7 Callback 75

3.8 PBX Advanced Settings 76

3.8.1 SIP settings 76

3.8.2 IAX Setting 82

3.8.3 PIN Sets 83

3.8.4 PIN Users 84

3.8.5 DISA 85

3.8.6 Paging and Intercom 87

3.9 Voice Management 88

3.9.1 System Recordings 88

3.9.2 Music on Hold 89

3.9.3 Voicemail Settings 91

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 3.9.4 System Prompts Settings 92

3.10 System Preferences 94

3.10.1 Firewall Rules 94

3.10.2 Security Info 96

3.10.3 Firmware update 97

3.10.4 Data Backup 99

3.10.5 Data Restore 99

3.10.6 Password 100

3.10.7 Time & Date 100

3.10.8 Reset 101

3.10.9 Reboot 101

3.11 Phone Provisioning 102

3.11.1 General Settings 102

3.11.2 Phones 103

3.12 Reports 104

3.12.1 CDR Report 104

3.12.2 System Logs 105

3.12.3 Firewall Logs 106

3.12.4 Trace Logs 106

3.13 System tools 108

3.13.1 SMTP Parameter 108

3.13.2 AMI Settings 109

3.13.3 Ping 110

3.13.4 Tracert 110

3.13.5 Packet Capture 111

3.13.6 Text to Wav 112

3.13.7 Certificates 112

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 1. Introduction
1.1 Overview

NEXO Series PBX​—IP PBX for Small Business/Home Office

NEXO 450/8100/16300 IP PBX is a standalone embedded hybrid PBX for
small businesses and remote branch offices of larger organizations. It is
designed to bring enterprise-grade Unified Communications and Security
Protection in an easy-to-manage fashion.

1.2 Product Features

● Alert ● Firewalls
● Blacklist ● HTTPS
● Call Back ● Integrated built-in packet
capture tools
● Call Detail Records(CDR) ● Interactive Voice Response (IVR)
● Call Forward,Call Parking ● Intercom/Zone Prompt
● Call Pickup ● Music On Hold
● Call Recording ● Open VPN
● Call Routing ● Paging/Intercom
● Call transfer ● Phone Provisioning
● Call Waiting ● PIN Users
● Caller ID ● QoS
● Conference ● Queue
● DDNS ● Ring Group
● Define Office Time ● Speed Dial
● Direct Inward System ● Spy functions
Access (DISA)
● Distinctive Ringtone ● Static Route
● Do Not Disturb(DND) ● VLAN
● External Storage ● Voicemail
●T.30,T.38 Faxes ●Alert Settings
●IP Blacklist ●AMI Settings
●Extension CDR
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 1.3 Product Appearance

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 Table 1-3-1 Description of Front view

Index Indicators Description

On: Starting
1 RUN Off: Abnormal
Blinking every 0.5s: Normal status
On: Power on
2 PWR
Off: Power off
Green LED: indicates the Internet interface is in Link .
3 WAN,LAN
Yellow LED: ON is indicates 100MBps Ethernet port.
Red LED s​ tands for FXO port
Orange LED i​ ndicates presence of a BRI port.
Green LED s​ tands for FXS port
Red L​ ED blinks: FXO port isn‟t connected to PSTN line.
Alternately blinks R
​ ed a​ nd ​Green​: FXO port has an
1~4,(5~8),
4 incoming call.
(9~16)
Alternately blinks R​ ed a​ nd ​Green f​ ast: FXO port is in a
call.
Alternately blinks G
​ reen a​ nd ​Red​: FXS port is ringing.
Alternately blinks G​ reen a​ nd ​Red f​ ast: FXS port is in a
call.

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 Table 1-3-2 Description of Rear view

Index Interface Description

Reset button to restore default IP and password or
restore factory setting.
1 RST
Hold RST button 8 seconds, RUN LED being ON during
this time
2 DC 12V Power connector of DC power. Input: DC12V 3A/DC12

1A (NEXO 450 only)

3 USB For the storage of call recording files

NEXO 8100 provides two 10/100 adaptive RJ45 Ethernet
ports, marked as LAN and WAN.
-LAN port :​ LAN port is for the connection to Local Area
WAN,LAN Network
4
-WAN port:​WAN port is the netword port for the
connection to internet. It supports “DHCP
server”,”PPPoE/dynamic DNS”,and”static IP”for IP
address assignment.
FXO port (​ red light): For the connection of PSTN lines
or FXS port of traditional PBX. NEXO 8100 users could
make or receive calls via FXO port.
FXS port (green light): For the connection of
1~4,(5~8), analog phones.
5
(9~16) BRI port(orange port): For the connection of ISDN
BRI lines. NEXO 8100 users could make or receive
calls via BRI port.
Note:The sequence number of the port corresponds
to that of the indicator lights in the front panel.

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 1.4 Scenario of Application
Application 1 - Figure 1.4.1

Application 2 - Figure 1.4.2

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 2. Installation Guide
2.1 Installation Notice
We use the NEXO 8100 device as an installation case as follows:

NEXO 8100 adapts 12VDC Power adapter, make sure AC power supply
grounded well to ensure the reliability and stability;
Notes: incorrect power connection may damage power adapter and device.
NEXO 8100 provides standard RJ45 with 10Mbps or 100Mbps interfaces.

2.2 Installation Procedure

2.2.1 Connect Drawing

Figure 2.2.1 Connect Drawing

3. WEB Interface Configuration

PBX IP PBX has the same web interface. This chapter describes web
configuration of PBX. The PBX contains an embedded web server to set
parameters by using the HTTP protocol. We are strongly recommend to
access device with Google Chrome or Firefox Browser.

We use the NEXO 8100 device as a configuration case as follows:

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 3.1 Access NEXO 8100 unit
Enter IP address of NEXO 8100 in IE/Google Chrome/Firefox Browser.
The default IP of LAN port is 192.168.6.200. and the GUI shows as
below: I​ n this example, the IP address is 192.168.6.91

Figure 3.1.1 WEB login interface

Enter username and password and then click “Login” in configuration interface.
The default username and password are “admin/admin”. It is strongly
recommended, change the default password to a new password for system
security .

3.2 Parameters Configuration

PBX WEB configuration interface consists of the navigation tree and the detail
configuration interfaces.

Figure 3.2.1 WEB introduction

Go through navigation tree, user can check, view, modify, and set the device
configuration on the right of configuration interface.
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 3.3 System Information
System information interface shows the basic information of status
information, mobile information and SIP information.

3.3.1 System Information

Figure 3.3.1 system Information

Table 3.3.1 System Information

Parameters Description
MAC Address Displays the current MAC of the gateway, for
example: 70-B3-D5-1B-3D-02
Network Current IP address and subnet mask of gateway
DNS Server Displays DNS server IP address in the same network with the
gateway
System Up Shows the time period of the device running. For
Time example, :1h : 20m : 24s
Traffic Calculates the net flow, including the total bytes of message
Statistics received and sent​。

Version info Shows the current firmware version

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 3.3.2 Extensions Status

Figure 3.3.2 Extensions Status

3.3.3 Trunk Status

Figure 3.3.3 Trunk Stratus

Trunk Status Description:

VoIP Trunk​:
Status
Rejected​: Trunk registration failed.
Registered​: Successful registration, trunk is ready for use.
Request Send​: Registering.
Waiting​: Waiting for authentication.
Service Provider​:
Status
OK​: Successful registration, trunk is ready for use.

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 Unreachable​: The trunk is unreachable.
Failed​: Trunk registration failed.
FXO Trunk​:
Status
Idle​: The port is idle.
Busy​: The port is in use.
Unavailable: The port hasn‟t connected to the PSTN line.
More detail message, please refer to the LED indication of front panel.

Table 3.3.3 Trunk Status

Parameters Description
Status Shows the registration status of Trunk channel, including
registered and unregistered.
Trunk Type Trunk mode will allow IP phone or IPPBX to register or trunk
mode to register to provider
Name It describes this VoIP channel for the ease of identification. Its
value is character string
SIP/IAX Choose the type of this trunk, SIP or IAX
Transfer This will be the transport method used by the trunk. The
Protocol options are UDP (default) or TCP or TLS.
User Name The number for this VoIP channel
Hostname/IP Hostname or IP Address of this VoIP channel
Address

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 3.4 Network Configuration
3.4.1 LAN Configuration

Figure 3.4.1 LAN Configuration

Table 3.4.1 Description of Local network

Parameter Description
Dynamic (DHCP) Enable the device obtain IP Address automatically
Static IP Address Configure the "IP Address", "Subnet Mask" and
"Default Gateway" by manual
Hostname Set the host name for PBX
IP Address Set the IP Address for PBX, It is recommended to
configure a static IP address for PBX
Subnet Mask Set the subnet mask for PBX
Gateway Set the gateway for PBX
IP Address 2 Set the second IP Address for PBX

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 Subnet Mask2 Set the second subnet mask for PBX
MTU Message transmit unit, default is 1500
Dynamic DNS Address Obtain DNS Server Address Automatically
Static DNS Address Obtain Primary DNS Server by manual
Primary DNS Server Set the primary DNS Server for PBX.
Secondary DNS Server Set the Secondary DNS Server for PBX.

Figure 3.4.1.2 WAN Configuration

Table 3.4.1.2 Description of WAN Configuration

Parameter Description
Use WAN Enalbe use wan

Dynamic (DHCP) Enable the device obtain IP Address automatically

Static IP Address Configure the "IP Address", "Subnet Mask" and "Default
Gateway" by manual
IP Address Set the IP Address for PBX, It is recommended to
configure a static IP address for PBX
Subnet Mask Set the subnet mask for PBX
Default Gateway Set the default gateway for PBX
Primary DNS Set the primary DNS Server for PBX.
Secondary DNS Set the Secondary DNS Server for PBX.
PPPoE Use PPPoE to a​ chieve IP address
User Name PPPoE user name
Password PPPoE password

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 3.4.2 VLAN Configuration

A VLAN (Virtual LAN) is a logical local area network (or LAN) that extends
beyond a single traditional LAN to a group of LAN segments, given specific
configurations.

Note: PBX is not the VLAN server, a 3-layer switch is still needed, please
configure the VLAN information there first, then input the details in PBX,
so that the packages via PBX will be added the VLAN label before sending
to that switch.

Figure 3.4.2 VLAN Configuration

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 Table 3.4.2 Description of VLAN Configuration

Parameter Description
NO.1 Click the NO.1 you can edit the first VLAN over LAN
IP Address Set the IP Address for PBX VLAN over LAN.
Subnet Mask Set the Subnet Mask for PBX VLAN over LAN.
Gateway Set the Default Gateway for PBX VLAN over LAN

3.4.3 ARP Configuration

The ARP function is mainly used to query and add the map of IP and MAC.
There are static or dynamic ARP entries.

Like other routers, the gateway can automatically find the network device on
the same segment. But, sometimes you don't want to use this automatic
mapping, you'd rather have fixed (static) associations between an IP address
and a MAC address. Gateway provides you the ability to add static ARP entries
to:
● Protect your network against ARP spoofing
● Prevent network confusion as a result of misconfigured network

device Click “Dynamic ARP” to check ARP buffer

Figure 3.4.3a Dynamic ARP

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 Figure 3.4.3 Add ARP

3.4.4 VPN Configuration

A Virtual Private Network (VPN) is a method of computer networking--typically

using the public internet--that allows users to privately share information
between remote locations, or between a remote location and a business' home
network. A VPN can provide secure information transport by authenticating
users, and encrypting data to prevent unauthorized persons from reading the
information transmitted. The VPN can be used to
send any kind of network traffic securely. PBX supports OpenVPN.
Figure 3.4.4 VPN Configuration

Table 3.4.4 Description of VPN Parameter

Parameters Description
Import VPN Import configuration file of OpenVPN.
Configuration Files

Notes:
1. Don't configure “user” and “group” in the “config” file. You can get the
config package from the OpenVPN provider.
2. PBX works as VPN client mode only.
3. Upload file *.tar with *.conf in it.

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 3.4.5 DDNS Server

DDNS(Dynamic DNS) is a method / protocol / network service that provides

the capability for a networked device, such as a router or computer system
using the Internet Protocol Suite, to notify a Domain Name System (DNS)
name server to change, in real time, the active DNS configuration of its
configured hostnames, addresses or other information.

Figure 3.4.5 DDNS Server

Table 3.4.5 Description of DDNS Server

Parameters Description
DDNS Server Select the DDNS server IP or domain name you sign up for
service.
User Name User name the DDNS server provides you.
Password User account‟s password.
Host Name The domain name you have got from the DDNS server

Note: DDNS allows you to access your network using domain names instead of
IP address. The service manages changing IP address and updates your
domain information dynamically. You must sign up for service through
dyndns.org, freedns.afraid.org, ​www.no-ip.com, w
​ ww.zoneedit.com

3.4.6 Static Route

PBX will have more than one internet connection in some situations but it has
only one default gateway. You will need to set some Static Route for PBX to
force it to go out through different gateway when access to different internet.
The default gateway priority of PBX from high to low is VPN/VLAN​-> ​LAN port.

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 1) Route Table

The current route rules of PBX.

Figure 3.4.6 Static Routing Table

2) Static Route Rules

You can add new static route rules here.

Figure 3.4.6a Static Routing Rules

Table 3.4.6 Description of Static Routing

Parameters Description
Destination The destination network to be accessed to by PBX.
IP Address
Subnet Mask Specify the destination network portion.
Gateway Define which gateway PBX will go through when access to the
destination network.
Metric The cost of a route is calculated by using what are called
routing metric. Routing metrics are assigned to routes by
routing protocols to provide measurable statistic which can be
used to judge how useful (how low cost) a route is.
Interface Define which internet port to go through.

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 3.4.7 DHCP Server

Figure 3.4.7 DHCP Server

Table 3.4.7 Description of DHCP Server

Parameters Description
Status DHCP service status

DHCP Enable Enable DHCP service

Start Address Start IP of DHCP IP pool

End Address End IP of DHCP IP pool

Default Lease Default lease time

Time
Gateway Gateway address

Subnet Mask Specify the destination network portion.

Address
Primary DNS Set the primary DNS Server for PBX.

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 Secondary Set the Secondary DNS Server for PBX.
DNS
Primary NTP Set the primary NTP Server
Server
Secondary Set the Secondary NTP Server
NTP Server
WINS Server Set the WINS Server Address
Address
TFTP Server Set the TFTP Server
Server
Allow Bootp Allow bootp clients
Clients

3.5 Trunks
3.5.1 Physical Trunks(PSTN and GSM Trunks)

The public switched telephone network (PSTN) is the network of the world's
public circuit-switched telephone networks.

Figure 3.5.1 Analog Trunks

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 Figure 3.5.1a Analog Trunks Edit

Table 3.5.1 Description of Analog Trunk

Parameters Description

Trunk Name A unique label used to identify this trunk when listed in
outbound rules, incoming rules, etc.E.g. “pstn113”.
Rxgain Used to modify the volume level of this trunk. Normally,
this setting does not need to be changed.
Answer on Polarity Use a polarity reversal to mark when a outgoing call is
Detection answered by the remote party
CID Detection For FXO trunks, this option forces PBX to look for Caller ID
on incoming calls.

CID Start This option allows you to define the start of a Caller ID
signal:
Ring: Start when a ring is received (Caller ID Signaling:

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 Bell_USA, DTMF).
Polarity: Start when a polarity reversal is started (Caller ID
Signaling: V23_UK, V23_JP,DTMF).
Before Ring: Start before a ring is received (Caller ID
Signaling: DTMF).
CID Signalling This option defines the type of Caller ID signaling to use.
It can be set to one of the following:
● Bell_USA: bell202 as used in the United States
● v23_UK: suitable in the UK
● v23_Japan: suitable in Japan
● v23-Japan pure: suitable in Japan
● DTMF: suitable in Denmark, Sweden, and Holland
Busy Detection Busy Detection is used to detect far end hang-up or for
detecting a busy signal. Select “Yes” to turn this
feature on.
Budy Count If Busy Detection is enabled, it is also possible to specify
how many busy tones to wait for before disconnecting the
call. The default is 4, but better results can be achieved if
set to 6 or even 8. Remember, the higher the number, the
more time will be required to release a channel. A higher
setting lowers the probability that you will encounter
random hang-ups.
Busy Interval The busy detection interval

Busy Pattern If Busy Detection is enabled, it is also possible to specify

the cadence of your busy signal.In many Countries, it is
500msec on, 500msec off. If a Busy Pattern is not
specified,The system will accept any regular sound-silence
pattern that repeats <Busy Count> times as a busy signal.
If you specify Busy Pattern, then the system will further
check the length of the tone and silence, which will further
reduce the chance of a false positive disconnection.
Frequency Used for Frequency Detection (Enable detecting the busy
Detection signal frequency or not).
Busy Frequency If the Frequency Detection is enabled, you must specify
the local frequency.
Hangup Polarity The call will be considered as “hang up” on a polarity
Reversal Detection reversal.

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 Figure 3.5.1b GSM Trunks

Figure 3.5.1c GSM Trunks Edit

Table 3.5.1c Description of GSM Trunk

Parameters Description

Port A port for this trunk.

Trunk Name A name for this trunk.
Mobile Number Mobile number for this trunk.
CLIR Calling Line Identification Restriction.
Rx Gain The receive volume.
Tx Gain The transfer volume.
Call Progress Tone A ringback for this trunk.
DTMF Detect Mode Set default dtmfmode for detect DTMF.
Default: Echo Before
Echo Before: Detect DTMF before echocan.
Echo After: Detect DTMF after echocan.
DTMF Detect DTMF detect sensitive.
Sensitive
PIN The PIN is normally associated with the SIM card.

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 3.5.2 IP Trunk (Peer to Peer Mode)

Figure 3.5.2 IP Trunk

Figure 3.5.2a Add IP Trunk

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 Figure 3.5.2b Add Bulk Dod

Table 3.5.2 Description of IP Trunk

Parameters Description
IP Trunk Add remote IP of Softswitch, SIP server which will send call
traffics to gateway.
Trunk Name It describes the trunk for the ease of identification.
Type Choose the type of this trunk, SIP or IAX
Outbound Caller ID for calls placed on out this trunk
Caller ID
Hostname/IP Service provider‟s hostname or IP address,5060 is the
Address standard port number used by SIP protocol. Don‟t change
this part if it is not required.
Transport This will be the transport method used by the SIP Trunk.
This method is given by the SIP trunk provider. The options
are UDP (default) or TCP or TLS.

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 DTMF Mode Set default mode for sending DTMF of this trunk. Default
setting: rfc2833, Info, Shortinfo,Inband, Auto
Qualify Send checking alive packets to the SIP provider. when it‟s
disabled, PBX will ignore the reachability and the status of
this account will be unmonitored.
Allow codecs ulaw,alaw,gsm
DOD Add dod number to associated extension.
Settintings
Add Bulk DOD Add bulk dod number to associated extensions which begin
with Begin number

3.5.3 VoIP Trunk

In this page, we can configure VoIP trunk (SIP/ IAX) you have got from
provider with the authorization name and password.

Figure 3.5.3 VoIP Trunk

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 Figure 3.5.3a Add VoIP Trunk

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 Figure 3.5.3b Add Bulk DOD

Table 3.5.3 Description of VoIP Trunk

Parameters Description
Trunk Name It describes the trunk for the ease of identification.
Type Choose the type of this trunk, SIP or IAX
Outbound Caller Caller ID for calls placed on out this trunk
ID
Hostname/IP Service provider‟s hostname or IP address, 5060 is the
Address standard port number used by SIP protocol. Don‟t change
this part if it is not required.
User Name User name of SIP account.
Password Password of SIP account.
Authorization Used for SIP authentication, it‟s the same as user name
Name generally.
Domain VoIP provider‟s server domain name

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 From User All outgoing calls from this SIP Trunk will use the From
User in From Header of the SIP Invite package. Keep this
field blank if it‟s not needed.
Transport This will be the transport method used by the extension.
The options are UDP (default) or TCP or TLS.
SRTP Define if SRTP is enabled for this trunk, it depends on
provider‟s configuration.
DTMF Mode RFC2833, Info, Shortinfo, Inband, Auto.
Qualify Send check alive packets to IP phones, when it‟s disabled,
PBX will ignore the reachability and the status of this
account will be unmonitored.
Allow codecs ulaw,alaw,gsm
Domain VoIP provider‟s server domain name
Proxy Address A proxy that receives requests from a client, even though
it may not be the server resolved by the Request-URI.
DOD Settintings Add dod number to associated extension.

Add Bulk DOD Add bulk dod number to associated extensions which begin
with Begin number

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 3.6 PBX Basic
3.6.1 Extensions

3.6.1.1 FXS Extensions

There are three types of extensions supported in PBX: SIP, IAX and analog
extension(FXS).
Figure 3.6.1.1 Extensions

Figure 3.6.1.1a Fxs Extensions Edit

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 Table 3.6.1.1a FXS Extensions

Parameters Description

Port The extension correspond port.

Extensions The numbered extension, e.g. 601, that will be associated
Number with this particular User/Phone.
Display Name A character-based name for this user, e.g. “Han Jones”.
Call ID The Caller ID (CID) string will be used when this user calls
another internal user.
Outbound CID Overrides the caller ID when dialing out a trunk.Any setting
here will override the common outbound caller ID set in
the trunks admin .
Format: “caller name” <######>
Leave this field blank to disable the outbound caller
ID feature for this user
Emergency CID This Caller ID will always be set when dialing out an
outbound route flagged as emergency.The emergency CID
overrides all other callerID Settings.

Figure 3.6.1.1b Fxs Extensions Voicemail

Table 3.6.1.1b Description of FXS Extensions Voicemail

Parameters Description

Enable Voicemail Check this box if the user should have a voicemail account.
Disable PIN Disable voicemail PIN authentication.
PIN Number Password used to access the Voicemail system.​e.g. “601”.

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 Email Address This option defines whether or not voicemails/Fax is sent to
Email Attachment the Email address as an attachment.
Note​: Please ensure that all voicemail settings are properly
configured on the System
Play CID Read back caller's telephone number prior to playing the incoming
message.
Play Envelope Envelope controls whether or nor the Voicemail system will
play the message envelope (date/time) before playing the
voicemail message.
Delete Voicemail the message will be deleted from the Voicemailbox (after having
been emailed).

Figure 3.6.1.1c FXS Extensions Options

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 Table 3.6.1.1c Description of FXS Extensions Options
Parameters Description

Call Forward This function sets inbound call forwarding on an extension.

(Follow Me) An administrator can configure Call Forward for this
extension.
Volume Settings Rxgain: The Volume sent to FXS extension.
Txgain: The Volume sent out by the FXS extension
Mobility Mobile Num: if you set a mobile number as mobility
Extension extension,while you call in PBX with this mobile number,
the mobile phone will get all permission of the associated
extension.for example: dialing the extension, playing the
voicemail.
Enable RingAll: when someone calls the associated
extension,your mobile phone will ring together,what you
need is set outbound route and set Outbound Prefix
number.
Maximum Call The absolute maximum amount of time permitted for a call,it only
valid for outbound calls
Duration
Ring Time Number of seconds to ring prior to going to voicemail.
Call Waiting Check this option if the extension should have Call Waiting
capability. If this option is checked, the “When busy” follow
me options will not be available.
Pinless Dialing Pinless Dialing will allow this extension to bypass any pin codes
normally required on outbound calls
Call Group Call group for peer/user
Pickup Group If this extension belongs to a pickup group, any calls that
ring this extension can be picked up by other extensions in
the same pickup group by dialing the Call Pickup feature
code(the default is *8).
Note​: *8 is the default setting, it can be changed under
Feature Codes -> General -> Call Pickup.
Do Not Disturb Do Not Disturb

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 Figure 3.6.1.1d Fxs Extensions Other

Table 3.6.1.1d Description of FXS Extensions Other

Parameters Description

Spy Settings PBX allows extension to monitor/barge in other

conversation. Once this feature is enabled, the extension
has the ability to monitor/barge in other calls using the
feature codes for each spy mode. Refer to “Feature Codes”
section
for more information.
spy modes There are 4 spy modes available:
● General spy: you have the permission to use the
following 3
modes.
● Quiet spy: you can only hear the call, but can't talk.
● Whisper spy: you can hear the call, and can talk with
the monitored extension.
● Barge spy: you can hear the call and talk with them
both.
Flash Sets the amount of time, in milliseconds, that must pass
since the last hook-flash event received by PBX before it will
recognize a second event. If a second event occurs in less
time than defined by Hook Flash Detection, then PBX will
ignore the event. The default value of Flash is 1000ms, and
it
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 can be configured in 1ms increments.
Web Login Extension web login setttings.

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 3.6.1.2 VoIP Extensions

A VoIP extension is a SIP/IAX Account that allows an IP Phone or an IP soft

phone client to register on PBX.
Figure 3.6.1.2 VoIP Extensions Edit/Add

Figure 3.6.1.2a VoIP Extensions Edit/Add

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 Table 3.6.1.2a Description of VoIP Extensions Edit/Add
Parameters Description

Extension Type Extension type: SIP, IAX or SIP/IAX.

● SIP​—The extension sends and receives calls using
the VoIP protocol SIP.
● IAX​—The extension sends and receives calls using the
VoIP protocol IAX.
Extension The numbered extension, e.g. 100, that will be associated
Number with this particular User/Phone.
Display Name A character-based name for this user, e.g. “Han Jones”.
Caller ID The Caller ID will be used when this user calls another
internal extension.
Outbound CID Overrides the caller ID when dialing out a trunk.Any setting
here will override the common outbound caller ID set in
the trunks admin .
Format: “caller name” <######>
Leave this field blank to disable the outbound caller
ID feature for this user
Emergency CID This Caller ID will always be set when dialing out an
outbound route flagged as emergency.The emergency CID
overrides all other callerID Settings.
Register Name It is for extension registration validation. Users will not be
able register the extension if the authorization name is
incorrect even though the username and password are
correct.
Password The password for this extension, but it is not a fixed one.
When you add new extension,a random and robust
password will be generated like “0e3lx9Iz”.

Transport This will be the transport method used by the extension.

The
options are UDP (default) or TCP or TLS.
SRTP Enable extension for SRTP (​RTP Encryption​).
DTMF Mode RFC2833, Info, Short Info,Inband, Auto.
Qualify Send check alive packets to IP phones.

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 NAT This setting should be used when the system is using a
public IP address to communicate with devices hidden
behind a NAT device (such as a broadband router). If you
have one-way audio problems, you usually have problems
with your NAT configuration or your firewall's support of SIP
and/or RTP ports.

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 Figure 3.6.1.2b VoIP Extensions Voicemail

Table 3.6.1.2b Description of VoIP Extensions Voicemail

Parameters Description

Enable Voicemail Check this box if the user should have a voicemail account.
Disable PIN Disable voicemail PIN authentication.
PIN Number Password used to access the Voicemail system.​e.g. “100”.
Email Address This option defines whether or not voicemails/Fax is sent to
Email Attachment the Email address as an attachment.
Note​: Please ensure that all voicemail settings are properly
configured on the System
Play CID Read back caller's telephone number prior to playing the incoming
message.
Play Envelope Envelope controls whether or nor the Voicemail system will
play the message envelope (date/time) before playing the
voicemail message.
Delete Voicemail the message will be deleted from the Voicemailbox (after having
been emailed).

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 Figure 3.6.1.2c VoIP Extensions Options

Table 3.6.1.2c Description of VoIP Extensions Options

Parameters Description

Call Forward This function sets inbound call forwarding on an extension.

(Follow Me) An administrator can configure Call Forward for this
extension.
Mobility Mobile N
​ um: if you set a mobile number as mobility
Extension extension,while you call in PBX with this mobile number,
the mobile phone will get all permission of the associated
extension.for example: dialing the extension, playing the
voicemail.
Enable RingAll: when someone calls the associated
extension,your mobile phone will ring together,what you
need is set outbound route and set Outbound Prefix
number.
Maximum Call The absolute maximum amount of time permitted for a call,it only
valid for outbound calls
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 Duration
Ring Time Number of seconds to ring prior to going to voicemail.
Call Waiting Check this option if the extension should have Call Waiting
capability. If this option is checked, the “When busy” follow
me options will not be available.
Allow Re-invite Re-Invite policy for this device.
yes: Allow RTP media direct.
no: Deny re-invites.
nonat: Allow reinvite when local, deny reinvite when NAT.
update: Use UPDATE instead of INVITE.
update,nonat: Use UPDATE when local, deny when NAT.",
Pinless Dialing Pinless Dialing will allow this extension to bypass any pin codes
normally required on outbound calls
Call Group Call group for peer/user
Pickup Group If this extension belongs to a pickup group, any calls that
ring this extension can be picked up by other extensions in
the same pickup group by dialing the Call Pickup feature
code(the default is *8).
Note​: *8 is the default setting, it can be changed under
Feature Codes -> General -> Call Pickup.
Do Not Disturb Do Not Disturb

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 Figure 3.6.1.2d VoIP Extensions Other

Table 3.6.1.2d Description of VoIP Extensions Other

Parameters Description

Spy Settings PBX allows extension to monitor/barge in other

conversation. Once this feature is enabled, the extension
has the ability to monitor/barge in other calls using the
feature codes for each spy mode. Refer to “Feature Codes”
section
for more information.
spy modes There are 4 spy modes available:
● General spy: you have the permission to use
the following 3 modes.
● Quiet spy: you can only hear the call, but can't talk.
● Whisper spy: you can hear the call, and can talk with
the monitored extension.
● Barge spy: you can hear the call and talk with them
both.

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 IP Restriction IP Restriction Settings
Default leave it blank on ”IP Restriction” configuration. it
indicate that registration of remote extension is
allowed(remote extension IP Address is not deny)

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 Deny: I​ P Address range to deny access to,in the form of
network/netmask, e.g.0.0.0.0/0.0.0.0
Permit: I​ P Address range to deny access to,in the form of
network/netmask,this can be a very useful security option
when dealing with remote extensions that are at a known
location(such as a branch office ) or within a known ISP
range for some home office situations.
e.g.192.168.6.1/255.255.255.0
Web Login Extension web login setttings.
Fax Associated Email: the email address that FAXs are send to.
Configuration It is used for T.38 FAX

3.6.2 Feature Codes

There are many feature codes available in PBX, which allow users to dial from
extension side to realize the exact feature.

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 Figure 3.6.2 Feature Codes

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 Table 3.6.2 Description of Feature Codes

Label Feature Codes Description

Call Pickup *8 Pickup extension

Trace last call number,and press 1,dial this

Call Trace *69
number out.
Directed Call [featurecode] + extension number
*08
Pickup Pickup specify extension
Attended [featurecode] + extension number
*2
Transfer Specify transfer to extension
[featurecode] + extension number
Blind Transfer ## After the success of the transfer to extension will
automatically hang up

One Touch Start recording in call, stop recording when Enter

*1
Record again
Call Forward All Call forward all activate
*72
Activate
Call Forward All Call forward all deactivate
*73
Deactivate
Call Forward Call forward busy activate
*90
Busy Activate
Call Forward Call forward busy deactivate
Busy *91
Deactivate
Call Forward Call forward no answer activate
No Answer *52
Activate
Call Forward Call forward no answer deactivate
No Answer *53
Deactivate
Call Forward to Call forward to voicemail
*900
Voicemail
Call Forward to Call forward to number
*901
Number
Call Forward Call forward to hang up
*902
Hang Up
Call Waiting - *70 Call waiting activate

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 Activate

Call Waiting -
*71 Call waiting deactivate
Deactivate

DND Activate *78 DND activate

DND
*79 DND deactivate
Deactivate
DND Toggle *76 DND toggle

Speed Dial [featurecode] + Speed Dial Source Number =

*0
Prefix Speed Dial Destination Number
Voicemail Main
*97 Into voicemail main menu
Menu
Dial Voicemail *98 Check extension voicemail

Direct Dial [featurecode] + Extension number

#
Prefix Leave a message to Specify extension

Call Parking *85 Eg. Park a call to extension 701

[featurecode] + Extension number

Quiet Mode *93
you can only hear the call, but can't talk.
[featurecode] + Extension number
Whisper Mode *94 you can hear the call, and can talk with the
monitored extension.

[featurecode] + Extension number

Barge Mode *95
you can hear the call and talk with them both.

Intercom Prefix *80 [featurecode] + Extension number

User Intercom
*54 Allow user intercom
Allow
User Intercom
*55 Disable user intercom
Disallow
[featurecode] + [password]
Access Code *99
Get into PIN Users function

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 3.6.3 Speed dial

Figure 3.6.3 Speed Dial

Figure 3.6.3a Speed Dial Add

Table 3.6.3 Description of Speed Dial

Parameters Description
Source Number The speed dial number.

Destination The number you want to call.

Number E.g. the source number is “33”. The destination number is
5528369. The prefix number is *90. You can use an
extension with any type to dial *9033, then it will call the
number 5528369.
The predix of Speed dial is setting on “feathur codes”
Note: D​ on‟t forget to add the outbound dial prefix if you
would like to dial the speed dial number through trunk.

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 3.6.4 Outbound Routes

In this page, we can configure the outbound rules to control the outgoing calls.
Notes:
1. The max number of outbound route is 32.
2. If the dial patterns are the same in several routes, PBX will
choose the available routes from top to the last one.
3. When you have created a new extension, please edit the outbound
route so that it can dial out too.

Figure 3.6.4 Outbound Routes

We can create outbound route or use the default route “9_outside” (dial
9+numbers to dial out). Also you can delete multiple outbound routes at once
as required.

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 Figure 3.6.4a Outbound Routes Edit

Table 3.6.4a Description of Outbound Routes Edit

Parameters Description

Route Name Name of this Outbound Route. E.g. “Local” or “Long

Distance”.

Route CID CID of this route

Override Whether ovrride extension cid

Extension
Route Passwd The route password can be used to protect this route from
being accessed without a password. You can choose one of

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 the passwords in the PIN list that you can click the “Pin
Settings” to edit it in “Pin Settings” page.
PIN SET Optional: Select a PIN Set to use.If using this option,Leave
the route password field blank.
Route Type ● Emergency
● Intra-Company
Disable All disable extensions
Extensions
Enable Define the extensions that will be permitted to use this
Extensions outbound route.
Disable Trunks All disable trunks
Enable Trunks Define the trunks that can be used for this outbound route.

Figure 3.6.4b Outbound Routes Edit

Table 3.6.4b Description of Outbound Routes Edit

Parameters Description
Prepend These digits will be prepended to the phone number before
the call is placed. For example,if a trunk requires 10-digit
dialing, but users are more comfortable with 7-digit dialing,
this field could be used to prepend a 3-digit area code to all
7-digit phone numbers before calls are placed.
Match Pattern Outbound calls that match this dial pattern will use this
outbound route. There are a number of dial pattern
characters that have special meanings:
X​: Any Digit from 0-9
Z​: Any Digit from 1-9
N​: Any Digit from
2-9
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 [12345-9] :​ Any digit in the brackets (in this example,
1,2,3,4,5,6,7,8,9)
The “​.​” Character will match any remaining digits. For
example, “9011.” will match any phone number that starts
with “9011”, excluding “9011” itself.
The “​!​” will match none remaining digits, and causes the
matching process to complete as soon as it can be
determined that no other matches are possible.
Example 1: 1
​ [5-8]6 w
​ ill match 156,166,176,186.
Example 2: 1 ​ NXXNXXXXX ​will match a phone number
starting with a 1, followed by a 3-digit area code, and
then
6-digit number.
Strip Allows the user to specify the number of digits that will be
stripped from the front of the phone number before the call
is placed. For example, if users must press 0 before dialing
a phone number, one digit should be stripped from the dial
string before the call is placed.
Add multiple dial patterns in this outbound route.
Add

Figure 3.6.4c Outbound Routes Edit

Table 3.6.4c Description of Outbound Routes Edit

Parameters Description
Office Hours When a specific office hour is selected, this outbound route
Mode can only be used during this office hour, and can‟t be used
in non-office hours.

Speciffic Office Configure specific office hour

Hours

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 3.6.5 Parking Lot

Figure 3.6.5 Parking Lot

Table 3.6.5 Description of Parking Lot

Parameters Description
Parking Lot This is the extension where you will transfer a call to park it.
Extension
Parking Lot The starting postion of the parking lot
Staring Postion
Number of Slots The total number of parking lot spaces to configure.
Example, if 700 is the extension and 8 slots are configured,
the parking slots will be 701-708
Parking Timeout The timeout period in seconds that a parked call will
(sec) attempt
to ring back the original parker if not answered(0 for 45s).
Alert Info This can create distinct rings on some SIP phones and can
serve to alert the recipients that the call is from an
Orphaned
parked call.

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 Parked Music This is the music class that will be played to a parked call
Class while in the parking lot UNLESS the call flow prior to parking
the call explicitly set a different music class, such as if the
call came in through a queue or ring group.
Transfer Enables or disables DTMF based transfers when picking up a
Capability parked call.
Re-Parking Enables or disables DTMF based parking when picking up a
Capability parked call.
Destination Destination to send the call to after Timeout Recording is
played.

3.6.6 Time Groups

Figure 3.6.6 Time groups configure

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 Figure 3.6.6a Time groups configure

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 3.6.7 General Preferences

Figure 3.6.7 General Preferences

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 Table 3.6.7 Description of General Preferences

Parameters Description
Select Language Web label language selection
English and Chinese-S
Max Account of Maximum concurrent calls limit(0 for unlimited)
Calls
Global Max Call The absolute maximum amount of time permitted for a call.
Duration A setting of 0 disables the timeout.
Ring Timeout Global extension ring timeout.

Country Please select your country or nearest neighboring country to

Tonezone enable the default dial tone, busy tone, and ring tone for
your region.

Muisc on Hold Select MOH music

Follow Me Play Mucic of follow me

Music on Hold Ring: normal ring back tone
Defaul: default MOH music
None: silence

FXO Mode FXO coutry mode

Feature Digigt Max time (ms) between digits for feature activation.
Timeout
Enable FTP FTP services, Default Port 21

Enable SSH SSH services, Default Port 8022

Enable HTTP HTTP services, Default Port 80

Enable HTTPS HTTPS services, Default Port 443

Extension The scope of VoIP Extension

Number
IVR Extensions The scope of IVR

Conference The scope of conference extension

Extensions
Queue The scope of queue extension
Extensions
Ring Group The scope of ring group
Extensions
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 Paging Group The scope of paging group
Extensinos

3.7 PBX Inbound Call Control

3.7.1 Inbound Routes

Inbound routing processes incoming call traffic to destination extensions

during office hours or outside office hours

Figure 3.7.1 Inbound Routes

There is a default inbound route for all the trunks and set IVR as the
destination, you can edit it or create a new one for your demands or you can
delete multiple outbound routes at once as required.When an incoming call
arrives, the system will first check “Holidays”.

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 Figure 3.7.1a Inbound Routes Edit

Table 3.7.1a Description of Inbound Routes Edit

Parameters Description
Route Name A name for this inbound route. E.g. “default”.

DID Number Define the expected DID Number if this trunk passes DID
on incoming calls. Leave this field blank to match calls with
any or no DID info. You can also use pattern matching to
match a range of numbers. The following patterns may be
used: ​X​: Any Digit from 0-9
Z​: Any Digit from 1-9
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 N​: Any Digit from 2-9
[12345-9]​: Any digit in the brackets (in this example, 1, 2,
3, 4, 5, 6, 7, 8, 9)
The “​.​” Character will match any remaining digits. For
example, “9011.” will match any phone number that starts
with “9011”, excluding “9011” itself.
The “​!​” will match none remaining digits, and causes the
matching process to complete as soon as it can be
determined that no other matches are possible.
Example 1: N
​ XXXXXX ​will match any 7-digit phone
number.
Example 2: 1
​ NXXNXXXXX ​will match a phone number
starting with a 1, followed by a 3-digit area code, and then
6-digit number.
Extension Define the extension for DID number. This field is only valid
when you use BRI, SIP, SPS or SPX trunk for this inbound
router. You can only input number and “-” in this field and
the format can be xxx or xxx-xxx. The count of the number
must be only one or equal to the count of the DID number.
Caller ID Define the Caller ID Number to be matched on incoming
Number calls. Leave this field blank to match any or no DID info.
You can also use a pattern match (e.g. 2[345]X) to match a
range of numbers.
The following patterns may be used:
X​: Any Digit from 0-9
Z​: Any Digit from 1-9
N​: Any Digit from
2-9
[12345-9]​: Any digit in the brackets (in this example, 1, 2,
3, 4, 5, 6, 7, 8, 9)
The “​.​” Character will match any remaining digits. For
example, “9011.” will match any phone number that starts
with “9011”, excluding “9011” itself.
The “​!​” will match none remaining digits, and causes the
matching process to complete as soon as it can be
determined that no other matches are possible.
Example 1: N
​ XXXXXX ​will match any 7 digits phone
number.
Example 2: 1
​ NXXNXXXXX ​will match a phone number

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 starting with a 1, followed by a 3-digit area code, and then
6-digit number.
Alert Info Alert info can be used for distinctive ring with SIP devices.

All Trunks List all available trunks

Allow Trunks This area allows you to select which trunks will be member
trunks for this route. To make a trunk a member of this
route, please move it to the “Selected” box.
Time Groups Select time groups mode.
Mode ● None: Disable office hours for this route.
● Gloal office hours: It is configured through
general preferences.
● Specific office hours: Use the specific office hours
settings.
Specific Time Set specific time groups
Groups
Day Destination ● End Calls
Route the incoming calls to end calls, the system will auto
NightDestination hang up the call.
● Extension
Route the incoming calls to a specific extension.
● Voicemail
Route the incoming calls to extension‟s voicemail.
● IVR
Route the incoming calls to a specific IVR.
● Ring Group
Route the incoming calls to a specific Ring Group.
● Conference Room
Route the incoming calls to a specific Conference Room.
● DISA
Route the incoming calls to a specific DISA.
● Queues
Route the incoming calls to a specific Queue.
● Outbound Routes
Route the incoming calls to a specific outbound route.
This function is mainly used for the connection of two
branches.
For example: Company A locates headquarters in the USA

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 with a branch B in China. A and B both have a PBX phone
system. Now if staff of A would like to make a call to a
telephone or mobile phone in China from the extension of A
but via the FXS line of B, that can be done by this
configuration.
Holiday Mode Define where the calls will be routed during Holidays.
● Select which defined Holiday to use.
● None: Disable holiday for this route.
● Gloal holiday: It is configured through
general preferences.
● Specific holiday: Use the holiday settings.
Specific Holiday Specific holiday time groups

Holiday Configure where to route the incoming calls during holidays.

Destination
Destination Fax detect destination

Figure 3.7.1b Inbound Routes Edit

Table 3.7.1b Description of Inbound Routes Edit

Parameters Description
CID Name Prefix Set inbound CID prefix

Signal RINGING Some devices or providers require RINGING to be sent

before ANSWER. You'll notice this happening if you can send
calls directly to a phone, but if you send it to an IVR, it
won't connect the call.
Enable Callback Enable callback

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 3.7.2 Blacklist

Blacklist is used to block an incoming/outgoing call. If the number of

incoming/outgoing call is listed in the number blacklist, the caller will hear the
following prompt: “The number you have dialed is not in service. Please check
the number and try again”. The system will then disconnect the call.

Figure 3.7.2 Blacklist

We can add a number to blacklist

Figure 3.7.2a Blacklist Add

3.7.3 IVR

When there‟s an inbound call aims at Auto Attendant, PBX will play an IVR
recording and route the caller to the requested destination (for example,
“Welcome to XX company,for sales press 1, for technical support press 2, for
operator press 0”, etc.). The system will transfer the call to corresponding
extension according to DTMF digits input by the user.

Figure 3.7.3 IVR

There is a default IVR here, we can edit it directly or add IVR by yourself.

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 Figure 3.7.3a IVR Add

Table 3.7.3a Description of IVR Add/Edit

Parameters Description

IVR Number PBX treats IVR as an extension; you can dial this extension
number to reach the IVR from internal extensions.
IVR Description Description of this IVR.

Announcement Greeting to be playd on entry to the IVR.

Enable Direct Allow the caller to dial other extensions number directly.
Dial
Timeout The number of times that the selected IVR prompt will be
played.
Invalid Retries Invalid retries number of keys

Invalid Destination when Number of times more than the settings.

Destination

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 Timeout Retries Retry timeout

Timeout Destination of timeout

Destination
CID Name IVR CID preifx name
Prefix
Key The Key pressed when the callers hear the IVR prompt.

Destination Where will PBX route the call when the action occurs.

Delete Delete a key to the destination IVR record.

Add Add a key to the destination IVR record.

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 3.7.4 Queue

Call Queues give users (e.g. call centers) an efficient means to have their calls
answeredin the order they were received to deliver top tier customer service.

Figure 3.7.4 Queue

Figure 3.7.4a Queue General

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 Table 3.7.4a Description of Queue General
Parameters Description

Queue Number Use this number to dial into the queue, or transfer callers to
this number to put them into the queue.
Queue Name A name for the Queue.
Queue You can require agents to enter a password before they can
Password log in to this queue.
Max Time Caller The maximum number of seconds a caller can wait in a
in Queue
queue before being pulled out (0 for unlimited).
Agents Timeout The number of seconds an agent's phone can ring before we
consider it a timeout.

CID Name CID preifx name

Prefix

Alert Info Alert info can be used for distinctive ring with SIP devices.

Ring Strategy This option sets the Ringing Strategy for this Queue. The
options are
● ringAll: Ring all available Agents simultaneously
until one answers.
● leastRecent: Ring the Agent which was least
recently called.
● fewestCalls: Ring the Agent with the fewest
completed calls.
● random: Ring a Random Agent.
● rrmemory: Round Robin with Memory,
Remembers where it left off in the last ring pass.
● Linear: Rings agents in the other specified,for
dynamic agents in the other they logged in.
Restrict Restrct dynamic agents
Dynamic Agents
Static Agents This selection shows all users. Selecting a user here makes
them an agent of the current queue.
Dynamic Agents Select dynamic agents

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 Figure 3.7.4b Queue Options

Table 3.7.4b Description of Queue Options

Parameters Description

Queue Weight Gives queues a 'weight' option, to ensure calls waiting in a

higher priority queue will deliver its calls first if there are
agents common to both queues.

Music on Hold Music (MoH) played to the caller while they wait in line for
Class an available agent.
Ringing Instead Enabling this option make callers hear a ringing tone instead
of Moh of Music on Hold.
Agent Announcement played to the Agent prior to bridging in the
Announcement caller
Join Announcement played to callers prior to joining the queue.
Announcement

Retry The number of seconds we wait before trying all the phones
again.
Warp-Up Time How many seconds after the completion of a call an Agent
will have before the Queue can ring them with a new call.(0
for no delay).
Ring In Use If set to no,the queue will avoid sending calls to members
whose devices are known to be 'in use'.

Report Hold If you wish to report the caller's hold time to the member
Time before they are connected to the caller, set this to yes.

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 Max Callers Maximum number of people waiting in the queue.

Join Empty This option controls whether callers can join a call queue
that has no agents. There are two options,
● Yes: Callers can join a call queue without agents or
only unavailable agents
● No: Callers cannot join a queue when there are no
agents in the queue.
Leave When This option controls whether callers already on hold are
Empty forced out of a queue that has no agents. There are two
options.
● Yes: Callers are forced out of a queue when no
agents are logged in.
● No: Callers will remain in a queue with no agents.

Figure 3.7.4c Queue Advanced Settings

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 Table 3.7.4c Description of Queue Advanced Settings
Parameters Description

Frequency How often to announce queue position and estimated hold

time.
Note​: “0 seconds” means disabling the announcement.

Announce Announce position of caller in the queue

Position
Announce Hold Enabling this option causes PBX to announce the hold time
Time to the caller periodically based on the frequency timer.
Either
yes or no; hold time will not be announced if <1 minute.
Prompt Select a prompt file to play periodically.
Frequency How often to announce a prompt to the caller.
Event When If a caller presses the key while waiting in the queue, this
Called setting selects which action should process the key press.
Member Status
Event
Service Level
Destination Define the failover action. A failover occurs after the user
reach the Queue max wait time.

3.7.5 Ring Groups

Ring groups can be configured to balance the call traffic for multiple users and
give callers a higher level of availability for incoming calls. Multiple ring
methods and voicemail are supported.
Note​: Call forward(follow me) feature in extension page will not take effect
when it‟s ringing as an agent.

Figure 3.7.5 Ring Grounps

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 Figure 3.7.5a Ring Groups Edit

Table 3.7.5a Description of Ring Groups Edit

Parameters Description

RG Number This option defines the numbered extension that can be

dialed to reach this group.

RG Name This option defines a name for this group, e.g. “Sales”.
“Ring Group Name” is a label to help you identify this group
in the group list.

Ring Strategy This option sets the Ringing Strategy for this Group. The
options are as follows:
● Ring All Simultaneously: Ring all available
Extensions simultaneously.
● Ring Sequentially: Ring each extension in the group one

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 at a time.
Ring Time 1. If the strategy is “Ring All Simultaneously”, it means
the number of seconds to ring this group before routing the
call according to the “Destination if No Answer” settings.
2. If the strategy is “Ring Sequentially”, it means the
number of seconds to ring a single extension before
moving onto the
next one.
Music on Hold If you select a music on hold class to play,instead of
“ring”,they will hear that instead ringing while they are
waiting for someone to pick up
Ring instead Of Enabling this option make callers hear a ringing tone instead
Moh of Music on Hold.

CID Name You can optionally prefix the caller ID name when ringing
Prefix
extensions in this group, ie: if you prefix with “Sales:”,a call
from John doe would display as “Sales:John doe” on the
extensions that ring.
Alert Info Alert info can be used for distinctive ring with SIP devices.

3.7.6 Conferences

Conference Calls increase employee efficiency and productivity, and provide a

more cost-effective way to hold meetings. Conference agents can dial * to
access the settings options and the admin can kick the last user out and can
lock the conference room.

Figure 3.7.6 Conferences

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 Figure 3.7.6a Conferences Edit/Add

Table 3.7.6a Description of Conferences Edit/Add

Parameters Description

Conference This is the number dialed to reach this Conference Room.

Number
Conference This option defines a name for this conference, e.g. “Sales”.
Name “Conference Name” is a label to help you identify this
conference in the conference list.

User PIN Set a PIN that must be entered in order to access this
conference room (e.g. 1234).
Admin PIN Enter a PIN number for the admin user
Join Prompt Message to be played to the caller before joining the
conference.

Max Paticipants Maximum Number of users allowed to join this conference.

Allow Menu Present Menu (user or admin) when '*' is received ('send' to
menu)

Music on Hold Enable Music On Hold when the conference has a single

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 caller.

Muisc on Hold Music (or Commercial) played to the caller while they wait in
Class line for the conference to start.

Quiet Mode Quiet mode (do not play enter/leave sounds)

User Count Announce user(s) count on joining conference

User join/leave Announce user join/leave

Leader Wait Wait until the conference leader (admin user) arrives before
starting the conference.

3.7.7 Callback

PBX allows caller A to dial an inbound route number, and after hearing the
ring, A can hang up the call or wait for PBX to cut off the call, then PBX will
call A with this number. When A picks up the call, A can dial the number he
wants to call; PBX will call the number with its outbound route.
Notes:
1. If you‟d like to use callback feature, please make sure it‟s enabled
on the inbound route setting panel.
2. No callback rules needed to be set if the trunk supports call back
with the caller ID directly.
Figure 3.7.7 Callback

Follow the steps below to use this function.

Step 1: Enable Callback.
Inbound Routes—Choose “Yes” on” Enable Callback” to enable this function.

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 Figure 3.7.7a Inbound route Callback settings

Step 2: Create Callback number.

Figure 3.7.7b Callback Edit/Add

3.8 PBX Advanced Settings

3.8.1 SIP settings

This is the SIP settings in PBX, including General settings, NAT, Codecs, Qos,
Response code and Advanced settings.
This section describes how to configure SIP server and SIP parameters

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 3.8.1.1 General

Figure 3.8.1.1 SIP General Setting

Table 3.8.1.1 Description of SIP General Setting

Parameters Description
Allowguest Whether allow anonymous registration extension.
Default: no. It‟s recommended to be disabled
for
security.
Allowoverlap Disable overlap dialing support.(Default is yes )
Pedantic Enable pedantic parameter. Default: no.

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 Always authreject If enabled, when PBX rejects “Register”or “Invite”
packets, PBX always respond the packets using
“SIP404
NOT FOUND”. It‟s recommended to be enabled for
security.
DNS SRV Look Up Please enable this option when your SIP trunk contains
more than one IP address.
Maxexpiry Maximum duration (in seconds) of a SIP
registration.Default is 3600 seconds.
Minexpiry Minimum duration (in seconds) of a SIP registration.
Default is 60 seconds.
Defaultexpiry Default Incoming/Outgoing Registration Time: Default
duration (in seconds) of incoming/outgoing registration.
Qualifyfreq How ofen to check for the host to be up in seconds and
reported in milliseconds with sip show settings.
Qualifygap Number of milliseconds between each group of peers
being qualified.
Register Timeout Number of seconds to wait for a response from a SIP
registrar before timed out. Default is 20 seconds.
Register Attempts The number of SIP REGISTER messages to send to a
SIP Registrar before giving up. Default is 0 (no limit).
RTPtimeout Terminate call if set # seconds of no RTP or RTCP activity
on the audio channel when we‟re not on hold.
RTPholdtimeout Both ends of the call time
RTPkeepalive Time of packaging
Notifyringing Control whether subscriptions already INUSE get send
RINGING when another call is sent.
Notifyhold Notify subscriptions on HOLD state.(default:no)
Session -timers Enable session-timer mode, default: yes. If you found
the call is cut off every 15 minutes every time,
please disable this.
Session-refresher Choose session-refresher, the default is Uas
Session-expires The max refresh interval
Session-minse The min refresh interval, which mustn't be shorter than
90s.
DTMF mode Set default mode for sending DTMF. Default setting:
rfc2833
Relaxdtmf Relax dtmf handing
Trustrpid If Remote-Party-ID should be trusted

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 Sendrpid If Remote-Party-ID should be sent
Contactdeny Use contactpermit and contactdeny to restrict at what IPs
Contactpermit your users may register their phones.
Canreinvite Asterisk by default tries to redirect the RTP media stream
to go directly from the caller to the callee.Some devices
do not support this (especially if one of them is behind a
NAT). The default setting is YES
Audioprefcodec Once enabled,When the caller call out via SIP/SPS
trunks,the audio codec of calling channel whould be
selected in preference.
usereqphone This provider requires,User=phone on URI
User agent To change the user agent parameter of asterisk, the
default is “PBX”, you can change it if needed.

3.8.1.2 Network

Note: Configuration of this section is required when using remote extensions

generally.

Figure 3.8.1.2 SIP Network Configuration

Table 3.8.1.2 Description of SIP Network Configuration

Parameters Description

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 Enable STUN STUN (Simple Traversal of UDP through NATs) is a
protocol for assisting devices behind a NAT firewall or
router with their packet routing.
STUN Address The STUN server allows clients to find out their public
address, the type of NAT they are behind and the
internet side port associated by the NAT with a
particular local port. This information is used to set
up
UDP communication between the client and the VOIP
provider and so establish a call.
External IP The IP address that will be associated with outbound SIP
Address messages if the system is in a NAT environment.
External Used to identify the local network using a network
Refresh Interval number/subnet mask pair when the system is behind a
NAT or firewall.
Some examples of this are as follows:
“192.168.0.0/255.255.0.0”: All RFC 1918 addresses are
local networks;
“10.0.0.0/255.0.0.0”: Also RFC1918;
“172.16.0.0/12”:Another RFC1918 with CIDR notation;
“169.254.0.0/255.255.0.0”: Zero conf local network.
Please refer to RFC1918 for more information.
External host Alternatively you can specify an external host, and the
system will perform DNS queries periodically. This setting
is only required when your public IP address is not
static. It is recommended that a static public IP address is
used with this system. Please contact your ISP for more
information.
NAT mode Global NAT configuration for the system; the options for
this setting are as follows:
Yes = Use NAT. Ignore address information in the
SIP/SDP headers and reply to the sender's IP
address/port.
No = Use NAT mode only according to RFC3581.
Never = Never attempt NAT mode or RFC3581 support.
Route = Use NAT but do not include report in headers.
RTP Port Start Beginning of RTP port range
UDP port Port used for SIP registrations, Default is 5060
TCP port Port used for SIP registrations, Default is 5060
TLS port Port used for SIP registrations, Default is 5061
TLS Verify Server When using PBX as a TLS client, whether or not to
verify server‟s certificate. It is “No” by default.
TLS Ignore Set this parameter as “No”, then common name must
Common Name be the same with IP or domain name.

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 TLS Verify Client When using PBX as a TLS server, whether or not to verify
client‟s certificate. It is “No” by default.
TLS Client Method When using PBX as TLS client, specify the
protocol for outbound TLS connections. You can select
it as tlsv1, sslv2 or sslv3.

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 3.8.1.3 Qos

Figure 3.8.1.3 Qos

3.8.1.4 Codecs

We can choose the allowed codec in PBX, a codec is a compression or

decompression algorithm that used in the transmission of voice packets over a
network or the Internet. More information about codec, you can refer to this
page: ​http://en.wikipedia.org/wiki/List_of_codecs

Figure 3.8.1.4 codecs

If you want to use codec G729, we recommend buying a license key and
input it here.

3.8.1.5 T.38

Figure 3.8.1.5 T.38

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 3.8.2 IAX Setting

IAX is the Internal Asterisk Exchange protocol, you can connect to PBX or
register IAX trunk to another IAX server. It’ is supported by the
asterisk-based IP PBX.
Figure 3.8.2 IAX setting

Table 3.8.2 Description of IAX setting

Parameters Description
Delayreject Which will delay the sending of authentication reject for
REGREQ or AUTHREP if there is a password
Bind port Port used for IAX2 registrations. Default is 4569.
Bandwidth Low/medium/high with this option you can control
which codec to be used.
Max Registration Maximum duration (in seconds) of an IAX2
Time registration. Default is 1300 seconds.
Min Registration Minimum duration (in seconds) of an IAX2 registration.
Time Default is 60 seconds.
Codec priority Codec priority controls the codec negotiation of an
inbound IAX call.This option is inherited to all user
entities
Codec Enable the codec you want for IAX communication.

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 3.8.3 PIN Sets

In this page users can manage all the passwords of outbound routes, PIN
User, and DISA.

Figure 3.8.3 PIN sets

Figure 3.8.3a PIN Set Edit

Table 3.8.3a Description of PIN Set Edit

Parameters Description
PIN Set Name A character-based name for this PIN list, e.g. “testPIN”
Record in CDR If set yes, the PIN code will be displayed in call log.
PIN List PIN list is a numeric field. Letters and punctuation are not
allowed in this field.
Fill in one PIN and if you end with enter for each PIN, you
could create multiple PINs.

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 3.8.4 PIN Users

Figure 3.8.4 PIN Users

Table 3.8.4 Description of PIN Users

Parameters Description
Authentication Number of times to retry when receiving an wrong
Retries password.
Digit Timeout The maximum amount of time permitted between digits
when the user is typing in an extension. Default of 5
seconds.
Join Waiting for validation, the system will play the prompt.
Announcement
Fail Announcement After validation fails, the system will play the prompt.

Figure 3.8.4a PIN Users Add/Edit

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 Table 3.8.4a Description of PIN Users Add/Edit
Parameters Description
PIN User Name A character-based name for this PIN list, e.g. “NEXOPIN”

PBX can store a number of PIN Users. PIN Users may be

used to keep track of calls in relation to particular
activities or clients. They can also be used to keep track
of calls by particular users or sets of users.
● PINs entered are checked against those stored by
the system. If an invalid PIN is entered, the PIN is
requested again.
● The system administrator can configure certain
numbers or types of numbers to require entry of a PIN
before users can continue making a call to such a
number.
● The system administrator can also configure to
require users to enter a PIN before making any
external
call.
Password The password for this PIN User.
PIN Set Click to add, delete or edit PIN list.
Allow Outbound PIN User can use those outbound route to make call out.
Routes

3.8.5 DISA

DISA (Direct Inward System Access) allows someone calling in from outside
the telephone switch (PBX) to obtain an “internal” system dial tone and make
calls as if they were using one of the extensions attached to the telephone
switch. To use DISA, a user calls a DISA number, which invokes the DISA
application. The DISA application in turn requires the user to enter a PIN
number, followed by the pound sign (#). If the PIN number is correct, the
user will hear dial tone on which a call may be placed. Obviously, this type
of access has serious security implications, and great care must be taken not to
compromise your security.

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 Figure 3.8.5 DISA

Figure 3.8.5 DISA Edit

Table 3.8.5 Description of DISA Edit

Parameters Description
DISA Name Give this DISA application a name to help you identify it.
Password The password for this DISA.
PIN Set Optional: select a PIN set to use.If using this option,leave
the password field blank
Response Timeout The maximum amount of time the system will wait before
hanging up the call if the user has dialed an incomplete
or invalid number. The default is 10 seconds.

Digit Timeout The maximum amount of time permitted between each

digit when the user is dialing an extension number. The
default is 5 seconds.
Caller ID (Optional) When using this DISA, the users CallerID will
be set to this. Format is "User Name" <5551234>.
Allow Outbound Used to set the outbound routes that can be accessed
Routes from this DISA.

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 3.8.6 Paging and Intercom

Paging is used to make an announcement over the speakerphone to a phone

or group of phones. Targeted phones will not ring, but instead answer
immediately into speakerphone mode. Please note that this section is for
configuring paging groups. If you would like to configure Intercom settings,
please open the PBX Basic -> Feature Codes screen.
Note​: A paging group can have a maximum of 20 members.

Figure 3.8.6 Paging and Intercom

Figure 3.8.6a Paging and Intercom Edit/Add

Parameters Description
Number Define the numbered extension that may be dialed to
reach this group.
Description The description of this paging group.
Force if Busy If selected, will not check if the device is in use before
paging it.
Duplex Paging is typically one way for announcements only.
Checking this will make paging duplex, allowing all users

in the group to talk and be heard by all.

Members Select members in this group.

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 3.9 Voice Management
3.9.1 System Recordings

We can record or upload the prompts in this page; you can also play it directly
to confirm if it‟s a valid one, you can also download it and save it as a backup.
Figure 3.9.1 Voice prompt Recording

1. Record New Prompt

Figure 3.9.1a Record New Prompt

The administrator can record custom prompts by doing the following:

1) Click “Record New Custom Prompt”.
2) Input the desired file name on the popup window and choose an
extension to call for recording (such as vp500).
3) Click “Record”. The selected extension will ring and you can pick
up the phone to start recording.

2. Upload

Prompt Click

“Upload”

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 Figure 3.9.1b Upload Voice Prompt

The administrator can also upload prompts by doing the following:

1) Click “Upload Prompt”.
2) Click “Browse” to choose the desired prompt.
3) Click “Save” to upload the selected prompt.

Note​: The file size must not be larger than 1.8 MB, and the file must be WAV format.

3.9.2 Music on Hold

Figure 3.9.2 Music on Hold

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 Figure 3.9.2a Music on Hold Edit

The administrator can upload on hold music as follows:

1) Click “Browse” to choose the desired audio file.
2) Click “Upload” to upload the selected file.

Note​: The file size must not be larger than 1.8 MB, and the file must be WAV
format:
GSM 6.10 8 kHz, Mono, 1 Kb/s;
Alaw/Ulaw 8 kHz, Mono, 1 Kb/s;
PCM 8 kHz, Mono, 16 Kb/s.

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 3.9.3 Voicemail Settings

In this page, we can configure some settings for voicemail feature, including
general voicemail settings and SMTP settings, which is used for “voicemail
to email”.

Figure 3.9.3 Voicemail Setting

Table 3.9.3 Description of Voicemail Setting

Parameters Description

Max # of Message Set the maximum number of messages that can be stored
Per Folder in a single voicemail box.

Max # of Login Max number of failed login attempts

Attempts
Max Length VM in Set the maximum length of a single voicemail message.

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 Sec
Min Length VM in Set the minimum length of a single voicemail message.
Sec Messages below this threshold will be automatically
deleted.
Max Length Max length of greeting in seconds.
Greetings in Sec
Review Message Allow sender to review/record their message before save
it(No by default)

From Email from

Email Subject Email subject
Email Body Email body
Play CID Say the called ID information before the message

Play Envelope Turn on/off envelope playback before message playback.

Say Duration Turn on/off the duration information before the message.

Move Message to Move heard messages to the “old” folder automatically

Old
Skip Message Ms Specifies how many milliseconds to skip forward/back
when the user skips forward or backward during message
playback.
Direct Dial to Default message type to use when dialing direct to an
Voicemail Message extensions voicemail
Type

Do Not Play Do Not Play "please leave message after tone" to Caller
"please leave
message after
tone" to Caller

3.9.4 System Prompts Settings

Upgrading of the system prompts package is possible through the

Administrator Web interface using a TFTP Server or an Upload
Enter your TFTP Server IP address and file location, then click start to update
the system prompts package

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 Figure 3.9.3 System Prompts Settings Upload

Figure 3.9.3a System Prompts Settings TFTP

Table 3.9.3 Description of System Prompts Settings

Parameters Description
File Name Choose a country voice package,filename must to '.tar.gz'
ending.
TFTP Server Tftp service server.

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 3.10 System Preferences
3.10.1 Firewall Rules

Figure 3.10.1 Firewall Rules

Figure 3.10.1a Firewall Rules Edit/Add

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 Table 3.10.1a Description of Firewall Rules

Parameters Description
Name A name for this rule. eg: HTTP.
Protocol The protocols for this rule.
IP The IP address for this rule. The format of IP address
is:IP/mask
● Ex:192.168.6.88/32 for ip 192.168.6.88
● Ex:192.168.6.0/24 for ip from
192.168.6.0 to 192.168.6.255
Port Initial port should be on the left and end port should be
on the right.The port must be equal to or greater than
start
port.
MAC The format of MAC Address is XX:XX:XX:XX:XX:XX, X
means 0~9 or A~F in hex, the A~F are not case
sensitive.
Target ● ACCEPT:Accept the access from remote hosts
● DROP:Drop the access from remote hosts
● REJECT:Reject the access from remote hosts

Figure 3.10.1b Firewall Defence Edit/Add

Table 3.10.1b Description of Firewall Defence Edit/Add

Parameters Description
Name A name for this rule. eg: HTTP.
Protocol The protocols for this rule.
IP The IP address for this rule. The format of IP address
is:IP/mask
● Ex:192.168.6.88/32 for ip 192.168.6.88
● Ex:192.168.6.0/24 for ip from
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 192.168.6.0 to 192.168.6.255
Port Initial port should be on the left and end port should be
on the right.The port must be equal to or greater than
start
port.
Limit Rate The maximum packets can be handled per unit time.
eg:(IP:192.168.6.88/32 Protocol:UDP Rate:10/sec)means
maximum 10 UDP packets from 192.168.6.88 can be
handled per minute,and drop the redundant packets.
Limit Hit The maximum connections can be handled per unit time.
eg:(Port:8022 Protocol:TCP Hit:10/minute)means
maximum 10 TCP connections to port 8022 can be
handled per minute,the eleventh connection will be
refused directly.

3.10.2 Security Info

Alert Settings, if the device is attacked, the system will notify users the alert
via call or E-mail. the attack modes include IP attack and Web Login.
Figure 3.10.2 Alert Settings

Figure 3.10.2a Alert Settings Edit

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 Table 3.10.2a Description of Alert Settings Edit
Parameters Description

Phone Notification Enable phone notification

Number Multiple extensions and outbound phone numbers could
be set for alert phone notification. Please separate
them by ';', e.g. '103;9XXX'.
Attempts The attempt times to dial a phone number when there is
no answer.
Interval The interval between each attempt to dial the phone
number.Must be greater than 3 seconds.
Prompt When answered, System will play this prompt.
Email Notification Enable email notification
To Multiple email addresses are allowed; please separate
them by ';', e.g. XXXX@gmail.com; ​YYYY@hotmail.com.
Email Subject Email subject
Email Body Email Body, Until 511 characters

IP Blacklist, if the device is attacked by IP attack.system will add ip to firewall

and Disable this IP access.

Figure 3.10.2b IP Blacklist

Table 3.10.2b Description of IP Blacklist

Parameters Description
Date IP Attack time
Protocol Attack protocol type
IP Attack ip
MAC Address Attack MAC address
Dest Port Attack destination port

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 3.10.3 Firmware update

Upgrading of the firmware is possible through the Administrator Web interface

using a TFTP Server or an Upload
Enter your TFTP Server IP address and firmware file location, then click start
to update the firmware

Notes:
1. If enabled “Reset configuration to Factory Defaults”, System will
restore to factory default settings.
2. When update the firmware, please don‟t turn off the power. Or the
system will get damaged.
Figure 3.10.3 Firmware Update Upload

Figure 3.10.3a Firmware Update TFTP

Table 3.10.3 Firmware Update

Parameters Description
Firmware update Send package file from your computer to the device
File name Firmware name,file must to „.img‟ ending.
Reset to Factory Reset Configuration to Factory Defaults
Setting
Browse Choose File

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 3.10.4 Data Backup

We can backup up the configurations before reset PBX to factory defaults

Figure 3.10.4

Click 'Backup' to download configuration file to your computer.

Notes:
1. Only configurations, custom prompts will be backed up.
2. When you have updated the firmware version, it‟s not
recommended to restore using old package.

3.10.5 Data Restore

You can restore this configuration in case the unit loses it for any reason or to
clone a unit with the configuration of another unit. The configuration backup
configurations are in txt format. Please note that you can use a backup file
from an older firmware version and use it in a unit with a more recent
firmware version. However, a backup file from a newer firmware version than
the one actually in the unit cannot be used for a restore operation on the unit.

Notes:
1. The upload process will last about 30s.
2. When you have updated the firmware version, it‟s not
recommended to restore using old package.

Figure 3.10.5

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 3.10.6 Password

When using web Configuration, please enter default user name and password.
User can modify the login name and password.

Figure 3.10.6 Password Setting

3.10.7 Time & Date

The Network Time Protocol (NTP) is a protocol and software

implementation for synchronizing the clocks of computer systems over
packet-switched, variable-latency data networks.
User need to fill the NTP Server Address and select Time Zone.

Figure 3.10.7 Time & Date parameter

Table 3.10.7 Time & Date parameter

Parameters Description
Time zone You can choose your time zone here.
Primary server Primary NTP Server Address

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 Secondary server Secondary NTP Server Address
Synchronism Set the time interval for checking local appliance‟s time
with the server
Daylight Saving Time Set the mode to Automatic or disabled
Manual Time Manual setup time

3.10.8 Reset

Be careful do this operation, after restore factory setting, all the parameters
will be changed to the factory default.

Figure 3.10.8 factory reset

Reset to Factory Defaults

Click this button to reset Factory Default settings

3.10.9 Reboot

Figure 3.10.9 Reboot

Warning: Rebooting the system will terminate all active calls!

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 3.11 Phone Provisioning
The Phone Provisioning provides users a method to Centralized config IP
Phone.

3.11.1 General Settings

Figure 3.11.1 General Settings

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 3.11.2 Phones

Figure 3.11.2 Configured Phones

Figure 3.11.2a Edit Phone

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 3.12 Reports
3.12.1 CDR Report

The call log captures all call details, including call time, caller number, callee
number, call type, call duration, etc. An administrator can search and filter call
data by call date, caller/callee, trunk, duration, billing duration, status, or
communication type.

Figure 3.12.1 CDR Report

Table 3.12.1 CDR Report

Parameters Description
Date start and end time of calls
Source Call number
Destination Called number
Src channel Source channel
Dst channel Destination channel
Call direction IP to GSM:
outbound calls from softswitch/IPPBX to mobile network
GSM to IP:
incoming calls from mobile network to IPPBX/Softswitch
Status Answered: the call was established successful
Canceled: the call was canceled by calling party
No Carrier: the call was rejected by mobile network
Not Answered: no body to answer the call
Busy: user busy
Duration Call duration of the call.

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 3.12.2 System Logs

Syslog is a standard for network device data logging. It allows separation of

the software that generates messages from the system that stores them and
the software that reports and analyzes them. It also provides devices which
would otherwise be unable to communicate a means to notify administrators
of problems or performance. There are 6 levels of syslog, including DEBUG,
NOTICE, WARNING and ERROR, EMERG,ALERT,CRIT,INFO.

Figure 3.12.2 System logs Local

Figure 3.12.2a System logs Server

Table 3.12.2 Description of System logs

Parameters Description
Export System Log ● Local: save log in local
● Server: save log in server
Log File Size Max size before rotation
Log File Count Rotated logs to keep (default: 4)

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 Syslog level Syslog Level
Server Address Server address
Server Port Server port

3.12.3 Firewall Logs

Figure 3.12.3 Firewall logs

Table 3.12.3 Description of Firewall logs

Parameters Description
Date IP Attack time
Protocol Attack protocol type
IP Attack ip
MAC Address Attack MAC address
Dest Port Attack destination port

3.12.4 Trace Logs

Figure 3.12.4 DAHDI Monitor Tool

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 Table 3.12.4 Description of DAHDI Monitor Tool

Parameters Description
Status Display recording status of using this tool.
Trunk Choose a Trunk to record.
Start Start recording
Stop Stop and download recordfile
Reset Reset recording and Cancel the recording file

Figure 3.12.4a Asterisk Logs

Table 3.12.4a Description of Asterisk Logs

Parameters Description
Enable Log Enable record asterisk log
Log File Size Log file size
Log File Count Rotated logs to keep (default: 8)
Log Level Asterisk log level
Enable SIP Debug Enable and set IP to enable sip debug
Enable RTP Debug Enable and set IP to enable rtp debug

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 3.13 System tools
3.13.1 SMTP Parameter

To send the SMS or system alert to email address, please configure the Email
settings first, and make sure SMTP test is successful.

Figure 3.13.1 SMTP Parameters

Table 3.13.1 SMTP Parameters

Parameters Description
Username The E-mail Address that PBX will use to send voice mail.
Password The password for the email address used above
SMTP Server The IP address or hostname of an SMTP server that the
PBX will connect to in order to send voice mail messages
via email, i.e.mail.yourcompany.com.
SSL If the server of sending email needs to authenticate the
sender, you need to enable this.
Note: Must be selected for Gmail or exchange server.
Port SMTP Port: the default value is 25.
Use SSL/TLS to If the server of sending email needs to authenticate the
send secure sender, you need to enable this.
message to server Note: Must be selected for Gmail or exchange server.

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 3.13.2 AMI Settings

The Asterisk Manager Interface(AMI) is a socket interface that you can use to
get configuration and status information, request actions to be performed,
and be notified about things happening to calls.

Figure 3.13.2 SMTP Parameters

Table 3.13.2 Description of SMTP Parameters

Parameters Description
Enable AMI Enable AMI setttins.

The Asterisk Manager Interface(AMI) is a socket interface

that you can use to get configuration and status
information, request actions to be performed, and be
notified about things happening to calls.
Username AMI user name,default “admin‟
Password AMI password,default “password‟
IP Restriction Set IP address and subnet mask that can connect to AMI

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 3.13.3 Ping

Figure 3.13.3 Ping

3.13.4 Tracert

Figure 3.13.4 Tracert

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 3.13.5 Packet Capture

Figure 3.13.5 Packet Capture

Table 3.13.5 Description of Packet Capture

Parameters Description
Status Packet capture status
Interface Choose network interface, LAN/WAN
Souce Capture souce Address
Destination Capture destination Address
Port Capture port
Protocol Capture protocol

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 3.13.6 Text to Wav

PBX can Transfer text to wav.

Figure 3.13.6 Text to wav

3.13.7 Certificates

PBX can support TLS trunk. Before you register TLS trunk to PBX, you should
upload certificates first.

Figure 3.13.7 Certificates

Trusted Certificate
This certificate is a CA certificate. When selecting “TLS Verify Client” as “Yes”,
you should upload a CA. The relevant IP PBX should also have this certificate.

Gateway Certificate
This certificate is server certificate. No matter selecting “TLS Verify Client” as
“Yes” or “NO”,you should upload this certificate to PBX. If IP PBX enables
“TLS Verify server”, you should also upload this certificate on IP PBX.

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