Yeastar S20, S50, S100, S300 Administrator's Manual

S-Series VoIP PBX

Administrator Guide

Sales Tel: +5411.4951.3104 #1 E-mail: ventas@rcmphones.com Support Tel: +5411.4951.3104 # 2 E-mail: soporte@rcmphones.com Web: http://www.rcmphones.com

Version: 30.1.0.10 Revised: 2016.09.01

S-Series VoIP PBX Administrator Guide

No parts of this publication may be reproduced or transmitted in any form or by any means, electronic or mechanical, photocopying, recording, or otherwise, for any purpose, without the express written permission of Yeastar Information Technology Co., Ltd. Under the law, reproducing includes translating into another language or format.

Declaration of Conformity

Hereby, Yeastar Information Technology Co., Ltd. declares that Yeastar SSeries IP PBX is in conformity with the essential requirements and other relevant provisions of the CE, FCC.

Warranty

The information in this document is subject to change without notice. Yeastar Information Technology Co., Ltd. makes no warranty of any kind with regard to this guide, including, but not limited to, the implied warranties of merchantability and fitness for a particular purpose. Yeastar Information Technology Co., Ltd. shall not be liable for errors contained herein nor for incidental or consequential damages in connection with the furnishing, performance or use of this guide.

WEEE Warning

In accordance with the requirements of council directive 2002/96/EC on Waste of Electrical and Electronic Equipment (WEEE), ensure that at end-of-life you separate this product from other waste and scrap and deliver to the WEEE collection system in your country for recycling.

S-Series VoIP PBX Administrator Guide

Contents

About This Guide ....................................................................................................................................... 5

S-Series Overview ...................................................................................................................................... 6

Introduction ............................................................................................................................................... 6

Feature Highlights .................................................................................................................................... 6

Expansion Board ...................................................................................................................................... 7

Hardware Overview .................................................................................................................................. 8

LED Indicators and Ports ....................................................................................................................... 10

Getting Started ......................................................................................................................................... 12

Accessing Web GUI ............................................................................................................................... 12

Web Configuration Desktop ................................................................................................................... 13

Make Your First Call ............................................................................................................................... 15

System Settings ....................................................................................................................................... 16

Network................................................................................................................................................... 16

Security ................................................................................................................................................... 20

User Permission ..................................................................................................................................... 25

Date & Time ............................................................................................................................................ 27

Email ....................................................................................................................................................... 28

Storage ................................................................................................................................................... 28

Extensions ................................................................................................................................................ 32

Add New Extension ................................................................................................................................ 32

Add Bulk Extensions............................................................................................................................... 37

Search and Edit Extensions ................................................................................................................... 38

Importing and Exporting Extensions ...................................................................................................... 38

Extension Group ..................................................................................................................................... 41

Trunks ........................................................................................................................................................ 42

FXO Trunk .............................................................................................................................................. 42

BRI Trunk ................................................................................................................................................ 44

GSM/3G Trunk ........................................................................................................................................ 47

VoIP Trunk .............................................................................................................................................. 48

E1/T1/J1 Trunk ....................................................................................................................................... 52

Call Control ............................................................................................................................................... 59

Inbound Routes ...................................................................................................................................... 59

Outbound Routes ................................................................................................................................... 61

Auto CLIP Routes ................................................................................................................................... 63

SLA ......................................................................................................................................................... 64

S-Series VoIP PBX Administrator Guide

Time Conditions ...................................................................................................................................... 66

Call Features ............................................................................................................................................. 68

IVR .......................................................................................................................................................... 68

Ring Group ............................................................................................................................................. 69

Queue ..................................................................................................................................................... 70

Conference ............................................................................................................................................. 72

Pickup Group .......................................................................................................................................... 73

Speed Dial .............................................................................................................................................. 74

Callback .................................................................................................................................................. 76

DISA ....................................................................................................................................................... 77

Blacklist/Whitelist .................................................................................................................................... 78

Pin List .................................................................................................................................................... 80

Paging/Intercom ..................................................................................................................................... 80

SMS ........................................................................................................................................................ 81

Voice Prompts........................................................................................................................................... 84

Prompt Preference ................................................................................................................................. 84

System Prompt ....................................................................................................................................... 84

Music on Hold ......................................................................................................................................... 85

Custom Prompt....................................................................................................................................... 86

General ...................................................................................................................................................... 88

Preference .............................................................................................................................................. 88

Feature Code .......................................................................................................................................... 89

Voicemail ................................................................................................................................................ 91

SIP .......................................................................................................................................................... 93

IAX ........................................................................................................................................................ 100

Recording ................................................................................................................................................ 101

Event Center ........................................................................................................................................... 102

Event Settings ...................................................................................................................................... 102

Notification Contacts............................................................................................................................. 102

Event Log.............................................................................................................................................. 103

CDR and Recording ............................................................................................................................... 105

PBX Monitor ............................................................................................................................................ 106

Extension Status................................................................................................................................... 106

Trunk Status ......................................................................................................................................... 107

Concurrent Call..................................................................................................................................... 109

Conference ........................................................................................................................................... 109

Resource Monitor ................................................................................................................................... 110

Information .............................................................................................................................................110

Network..................................................................................................................................................110

S-Series VoIP PBX Administrator Guide

Performance ..........................................................................................................................................110

Storage Usage.......................................................................................................................................112

Maintenance ............................................................................................................................................ 113

Upgrade .................................................................................................................................................113

Backup and Restore ..............................................................................................................................115

Reset and Reboot..................................................................................................................................117

System Log ............................................................................................................................................117

Operation Log ........................................................................................................................................117

Troubleshooting .....................................................................................................................................118

App Center .............................................................................................................................................. 121

What App Center Offers ....................................................................................................................... 121

Install Apps ........................................................................................................................................... 122

Manage Apps ........................................................................................................................................ 122

S-Series VoIP PBX Administrator Guide

About This Guide

Thanks for choosing Yeastar S-Series VoIP PBX. This guide is intended for administrators who need to prepare for, configure and operate S-Series IP PBX. In this guide, we describe every detail on the functionality and configuration of the PBX. We begin by assuming that you are interested in S-

Series VoIP PBX and familiar with networking and other IT disciplines.

Products Covered

This guide explains how to configure the following products:

 Yeastar S20 VoIP PBX  Yeastar S50 VoIP PBX  Yeastar S100 VoIP PBX  Yeastar S300 VoIP PBX

Related Documents

The following related documents are available on Yeastar website: http://www.yeastar.com.

Document

Description

Yeastar S-Series Datasheet

Datasheet for the Yeastar S-Series.

Yeastar S20 Installation Guide Yeastar S50 Installation Guide Yeastar S100 Installation Guide Yeastar S300 Installation Guide

Installation guide for the Yeastar Series IP PBX. Yeastar S-Series Extension User Guide

Users could refer to the manual for instructions on how to login the user portal, and how to configure their accounts, listen to call recordings, check voicemail messages, etc.

Safety when working with electricity

 Do not use a 3rd party power adaptor.  Do not power on the device during the installation.  Do not work on the device, connect or disconnect cables when lightning strikes.

S-Series VoIP PBX Administrator Guide

S-Series Overview

This chapter provides the following sections:

 Introduction  Feature Highlights  Expansion Board  Hardware Overview

Introduction

Designed with the small and medium sized enterprises in mind, supporting up to 500 users and built using the very latest technology, the Yeastar S-Series delivers exceptional cost savings, productivity and efficiency improvements, delivering power, performance, quality and peace of mind.

The all new S-Series is engineered for the communications needs of today and tomorrow, and with the Yeastar unique modular design future proofs your investment choice.

Feature Highlights

Appreciate the Easy-to-use Solution

 Intuitive and graphical UI brings point-and-click configuration.  Convenient Phone Provisioning feature saves you tremendous time.  Everything can be managed from anywhere with Internet access.

Your Choice of Technologies and Features

 Embedded VoIP capability and analog phone connections.  Rich external lines options include SIP, PSTN, ISDN BRI, E1/T1/PRI, and cellular networks.  Concurrent calls and maximum users are expandable with modules.  App Center integrates features that you can add when you need them.

Telephone System without Risk

 Meanwell power supply featuring MTBF>560Kh.  High-quality Freescale CPU processor and industry leading TI DSP voice processor.  Connectors from TE Connectivity with a gold plating layer as thick as 15 μ.  Lightening protection on analog ports complying with ITU-T K.20/45/21 8/20 μs and GR-1089

standard.

Play Safe and Expect Reliability

 TLS, SRTP, and HTTPS standards for better security.  Defend against malicious attack with built-in Firewall.  Monitor system status and behavior and be notified when abnormalities occur.

Learn more about Yeastar S-Series here:

http://www.yeastar.com/S_Series_VoIP_PBX

S-Series VoIP PBX Administrator Guide

Expansion Board

Yeastar S100 and S300 are expandable.

 S100 supports up to 2 EX08/EX30 Expansion Spans; supports 1 D30 Module.  S300 supports up to 3 EX08/EX30 Expansion Spans; supports up to 2 D30 Modules.

Expansion Board – EX08

EX08 board supports up to 4 modules (8 ports).

Expansion Board – EX30

EX30 board supports 1 E1/T1 port.

D30 Module

D30 is a DSP module, used to expand the capacity of PBX. With per D30 module added, the extensions increase 100 and concurrent calls increase 30 in additional.

Optional Module

 O2 Module  S2 Module  SO Module  B2 Module  GSM Module  3G Module

S-Series VoIP PBX Administrator Guide

Hardware Overview

Yeastar S20

Yeastar S50

Front Panel

Power Indicator System Status RJ11 Port Status WAN Status

LAN Status

Rear Panel

Power RJ11 Port WAN LAN Reset Antenna Socket TF Slot

Front Panel

System Indicator Power Indicator RJ11 Port Reset

WAN LAN RJ11 Port Status SD Slot

S-Series VoIP PBX Administrator Guide

Yeastar S100

Rear Panel

Power Switch Antenna Socket Power Inlet

Protective Earth

Front Panel (1*EX30 + 1*EX08)

E1/T1 Port RJ11 Port RJ11 Port Status

Rear Panel Antenna Socket

SD Slot Console Power Switch Power LAN Power Inlet

System WAN Protective Earth Reset USB Slot

S-Series VoIP PBX Administrator Guide

Yeastar S300

LED Indicators and Ports

LED Indicators

LED

Indication

Status

Description

POWER

Power status On

The power is switched on

Off

The power is switched off

System

System status Blinking

The system is running properly

Static/Off

The system goes wrong

WAN

WAN status Static Green light

Linked normally, 10/100 Mbps.

Static Orange light

Linked normally, 1000 Mbps.

Blinking

In communication.

Off

Off-line.

LAN

LAN status Static Green light

Linked normally, 10/100 Mbps.

Static Orange light

Linked normally, 1000 Mbps.

Blinking

In communication.

Off

Off-line.

Front Panel (1*EX30 + 2*EX08)

E1/T1 Port

RJ11 Port RJ11 Port

RJ11 Port Status RJ11 Port Status

Rear Panel Antenna Socket

SD Slot Console Power Switch Power LAN Power Inlet

System WAN Protective Earth Reset USB Slot

S-Series VoIP PBX Administrator Guide

RJ11 Port Status

FXS

Green light

 Static: The port is idle.  Blinking: There is an ongoing call on the

port.

GSM/3G

Red light

 Static: the trunk is idle.  Blinking slowly: there is no SIM card

inserted.

 Blinking rapidly: the trunk is in use.

BRI

Orange light

 Blinking slowly: the BRI line is

disconnected.

 Static: the BRI line is connected or in

use.

FXO

Red light

 Blinking slowly: no PSTN line is

connected to the port.

 Static: the PSTN line is idle.  Blinking rapidly: the PSTN line is busy.

Port Description

Ports

Description

RJ11 Port

FXO port (red light): for the connection of PSTN lines or FXS ports of traditional PBX.

FXS port (green light): for the connection of analog phones. BRI port (orange light): for the connection of ISDN BRI lines.

Note: the sequence number of the ports corresponds to that of the Indicator

lights in the front panel. (I.e. the LED lights in the front indicate the connection status of the corresponding ports at the front panel.)

ANT

Connect to GSM/3G Antenna.

E1/T1

Connect to E1 line or the E1 port of traditional PBX.

Console

Connect to the RS-232 Cable to debug to system.

TF Slot

Insert TF card.

SD Slot

Insert SD card.

USB Slot

Connect to USB external disk.

Ethernet Port

Yeastar S20 provides two 10/100M adaptive RJ45 Ethernet ports, S50/100/300 supports two 10/100/1000M Ethernet ports. There are 2 Ethernet modes for the system. The default mode is “Bridge”.  Bridge: LAN port interface will be used for uplink connection. WAN port

interface will be used as bridge for PC connection.

 Dual: both ports can be used for uplink connection.

Reset Button

Press and hold for 10 seconds to restore the factory defaults

Power Inlet

Connect the supplied power supply to the port.

Power Switch

Press this button to switch on/off the device.

S-Series VoIP PBX Administrator Guide

Getting Started

This chapter explains how to log in Yeastar S-Series Web GUI, use the taskbar and widgets, and open applications with the Main Menu.

 Accessing Web GUI  Web Configuration Desktop  Make Your First Call

Accessing Web GUI

Yeastar S-Series provides web-based configuration interface for administrator and extension users. The administrator can manage the device by logging in the Web interface. Check the factory defaults below: IP address: https://192.168.5.150:8088 User Name: admin Default Password: password

To log in S-Series: 1 Make sure your computer is connected to the same network as the IP PBX. 2 Start a web browser on your PC, enter the IP address, press Enter on your keyboard. 3 Enter your user name and password, click Login.

Figure 2-1 S-Series Web Configuration Panel Login Page

Note: To ensure your connection to the S-Series Web GUI runs smoothly, please use the following

browsers:

 Chrome  Firefox  Internet Explorer: 11.0 or later

S-Series VoIP PBX Administrator Guide

Web Configuration Desktop

When you log in Yeastar S-Series Web GUI, you will see the desktop. From here, you can manage settings, install applications, or view system resource information.

Desktop

The desktop is where your application windows are displayed.

Figure 2-2 Desktop

Taskbar

The taskbar at the top of the desktop includes the following items:

Figure 2-3 Taskbar

1 Main Menu: view and open applications installed on your S-Series system. Right-click an

application icon, you can add the application to desktop.

2 Open Application

 Click the icon of an application to show or hide its window on the desktop.  Right-click the icon and choose from the shortcut menu to manage the application window

(Maximize, Minimize, Restore, Close).

3 Notifications: displays notifications, like errors, status updates, and app installation notifications. 4 Resource Monitor: click the icon to check the system information, network status and storage usage. 5 Options: logout, change Web language or modify personal account options.

Main Menu

Click the Main Menu at the top-left of the desktop, you can find all the installed applications on your S-Series system.

S-Series VoIP PBX Administrator Guide

Figure 2-4 Main Menu

Options

Click the options icon to logout, change Web language or modify your account settings.

Figure 2-5 Options

 Language

Select Language to change web language.

 My Settings

Click My Settings to modify your account settings. Here you can change the login password and bind your email address with the account.

S-Series VoIP PBX Administrator Guide

Figure 2-6 My Settings

 Logout

Click Logout to log out the Web GUI.

Save and Apply Changes

Click Save button after your configurations on the S-Series system, do not forget to click Apply button on the upper right of the desktop to submit all the changes. If the change requires reboot to take effect, the system will prompt you with a pop-up window.

Make Your First Call

Connect your IP phone and S-Series device to the same network. Then register an extension to the IP phone and make your first call through S-Series system.

1 Log in your S-Series Web GUI, go to Settings > PBX > Extensions. 2 Click Add to create a new extension, set the type as “SIP”. You will need the Registration Name

and Registration Password to register the extension later.

3 Register the extension on your phone with the Registration Name and Registration Password,

the SIP server address is your S-Series IP address.

4 When the extensions is registered to S-Series, you can dial *2 to access your voicemail box. The

default password to enter the voicemail box is your extension number.

5 Once entering the voicemail box, you are connected to the S-Series system!

S-Series VoIP PBX Administrator Guide

System Settings

This chapter explains system settings on S-Series. Go to Settings > System to check the system settings.

 Network  Security  User Permission  Date & Time  Email  Storage

Network

After successfully logging in the S-Series Web GUI for the first time with the factory IP address, users could go to Settings > System > Network to configure the network for S-Series.

Yeastar S-Series supports 3 Ethernet modes: Single, Dual and Bridge.

Basic Settings

Please check the basic network settings below.

Table 3-1 Network Basic Settings Description

Basic Settings

Hostname

Set the hostname for the system.

Mode

Select the Ethernet mode. The default mode is Single.

 Single: only LAN port will be used for uplink, WAN port is disabled.  Bridge: LAN port interface will be used for uplink connection. WAN

port interface will be used as bridge for PC connection.

 Dual: the two Ethernet interfaces will use different IP addresses.

Assign two IP addresses in this mode.

Default Interface

In Dual mode, you need to choose the default interface.

LAN/WAN Settings (DHCP Mode)

If you choose this mode, the system will act as DHCP client to get an available IP address from your local network.

LAN/WAN Settings (Static IP Address)

IP Address

Enter the IP address (xxx.xxx.xxx.xxx).

Subnet Mask

Enter the subnet mask (xxx.xxx.xxx.xxx). For example, 255.255.255.0

Gateway

Enter the gateway address (xxx.xxx.xxx.xxx).

Preferred DNS Server

Enter the IP address of the preferred DNS server (xxx.xxx.xxx.xxx).

Alternate DNS Server

Enter the IP address of the alternative DNS server (xxx.xxx.xxx.xxx).

LAN/WAN Settings (PPPoE)

Username

Enter the PPPoE username.

S-Series VoIP PBX Administrator Guide

Password

Enter the PPPoE password.

VLAN

Enable VLAN

Check this option to enable VLAN.

VLAN ID

Enter the VLAN ID.

VLAN Priority

Set the VLAN priority. The default is 0.

OpenVPN

S-Series supports OpenVPN. The system provides detailed VPN configurations on the Web GUI and you can also upload the VPN configuration package to the system to make it work.

Before using OpenVPN feature, please Enable OpenVPN first, then choose the Type to configure OpenVPN:

 Manual Configuration  Upload OpenVPN Package

Check the VPN configurations parameters below.

Table 3-2 OpenVPN Manual Configuration Parameters Description

OpenVPN Configuration

Server Address

Enter the server address of OpenVPN.

Server Port

Enter the server port of OpenVPN. The default is 1194.

Protocol

Select the protocol type. The server and client must use the same protocol.

Device

Select the network device. The client and server must use the same setting.

 TUN: a TUN device is a virtual point-to-point IP link.  TAP: a TAP device is a virtual Ethernet adapter.

Username

Specify the username.

Password

Specify the password.

Encryption

Select the encryption method. The server and client must use the same setting.

Compression

Enable or disable compression for data stream. The server and client must use the same setting.

Proxy Server

Specify the proxy server.

Proxy Port

Specify the proxy port.

CA Cert

Upload a CA certificate.

Cert

Upload a Client certificate.

Key

Upload a Client key.

TLS Authentication

Enable or disable TLS authentication. If enabled, please upload a TA key via Settings > System> Security>Certificate.

S-Series VoIP PBX Administrator Guide

DDNS Settings

Dynamic DNS or DDNS is a method of updating, in real time, a Domain Name System (DNS) to point to a changing IP address on the Internet. This is used to provide a persistent domain name for a resource that may change location on the network. DDNS is usually configured on router. If your router cannot support DDNS, we can set up DDNS on Yeastar system.

Yeastar S-Series supports the following DDNS servers:

 dyndns.org  freedns.afraid.org  www.no-ip.com  www.zoneedit.com  www.oray.com  3322.org

Check the DDNS configuration parameters below.

Table 3-3 DDNS Configuration Parameters Description

DDNS

DDNS Status

This shows the current DDNS status of the device.

Enable DDNS

Check this box to enable DDNS.

Server

Choose a DDNS provider from the list.

Username

Enter the username of your DDNS account.

Password

Enter the password of you DDNS account.

Hash

Enter your string of Hash as provided by freedns.afraid.org.

Domain

Enter the domain name.

Static Route

In computer networking, a routing table is a data table stored in a router or a networked device that lists the routes to particular network destinations, and in some cases, metrics (distances) associated with those routes. Static routes are entries made in a routing table by non-automatic means and which are fixed rather than being the result of some network topology “discovery” procedure. Static route on the system is used to configure to route the connection, packets to particular network destinations, usually a specific gateway.

 Routing Table

All the static routes are displayed on the Routing Table.

S-Series VoIP PBX Administrator Guide

Figure 3-1 Routing Table

 Static Routes

Click Static Routes tab, you can add static routes here. Click to add a static route.

 Click to edit the static route.

 Click to delete the static route.

Check the Static route settings below.

Table 3-4 Static Routes Settings Description

Static Route

Destination

Enter the destination IP address or IP subnet for the S-Series to reach using the static route.

Example:

 IP address: 192.168.6.120  IP subnet: 192.168.6.0

Subnet Mask

Enter the subnet mask for the destination address.

Example:

255.255.255.255

Gateway

Enter the gateway address. The S-Series system will reach the destination address via this gateway.

Example:

192.168.6.1

Metric

The cost of a route is calculated using what are called routing metric. Routing metrics are assigned to routes by routing protocols to provide measurable values that can be used to judge how useful (how cost) a route will be.

Interface

Select the network interface. The system will reach the destination address using the static route through the selected

S-Series VoIP PBX Administrator Guide

network interface.

Security

VoIP attack, although not an everyday occurrence does exist. When using VoIP, system security is undoubtedly one of the issues we care about most. With appropriate configuration, and some basic safety habits, we can improve the security of the telephone system. Moreover, the powerful built-in firewall function in Yeastar system is adequate to enable the system to run safely and stably. We strongly recommend that you configure firewall and other security options to prevent the attack fraud and the system failure or calls loss.

Firewall Rules

Users could add rules to accept or reject traffic through the system. Go to Settings > System >

Security > Firewall Rules to configure firewall for the system.

Before adding firewall rules, please check the option Enable Firewall, then click Save to enable the firewall.

Figure 3-2 Firewall Rules

 Click to add a new rule.

 Click to edit the rule.

 Click to delete the rule.

Check the firewall configuration parameters below.

Table 3-5 Firewall Configuration Parameters Description

Firewall

Enable Firewall

Enable Firewall to protect the system from malicious attack. Click Save icon to apply the changes.

Disable Ping

Enable this item, net ping from remote hosts will be dropped. Click Save icon to apply the changes.

Drop All

When you enable Drop All feature, the system will drop all packets and connections from other hosts if there are no other rules defined. To avoid locking the device, at least one TCP Accept common rule must be created for port used for SSH access and port used for HTTP access.

Firewall Rules

Name

Specify a name to identify the firewall rule.

Description

Description for this firewall rule.

S-Series VoIP PBX Administrator Guide

Action

Select the action for the firewall rule:

 Accept  Ignore  Reject

Protocol

Select the protocol applied for the rule:

 UDP  TCP  BOTH

Source IP address/ Subnet mask

The IP address for this rule.

Example:

192.168.5.100/255.255.255.255 means this rule is for 192.168.5.100.

192.168.5.100/255.255.255.0 is for IP from 192.168.5.0 to 192.168.5.100.

Port

Set the port for the firewall rule. The end port must be equal to or greater than start port.

IP Auto Defense

Users could create auto defense rules, then the system will prevent massive connection attempts or brute force attacks. The IP addresses would be listed in the Blocked IP Address table. There are 3 default auto defense rules, we recommend you keep the rules there.

Figure 3-3 Auto Defense Rules

Please check the auto defense rule configuration parameters below.

Table 3-6 IP Auto Defense Rule Configuration

IP Auto Defense Rule

Port

Auto defense port, for example, 8022.

Protocol

Select auto defense protocol:

 UDP  TCP

The Number of IP Packets

The number of IP Packets permitted within a specific time interval. Time Interval

The time interval to receive IP Packets. For example, Number of IP Packets sets 90 and Time Interval sets 60 mean 90 IP packets are allowed

S-Series VoIP PBX Administrator Guide

in 60 seconds.

Service

The service page displays all the service status and port on S-Series.

Table 3-7 Service Configuration

Protocol or Service

Description

HTTPS

The default access protocol is HTTPS and the port is 8088.

Redirect from port 80

If the option is enabled, when you access S-Series using HTTP with port 80, it will be redirected to HTTPS with port 8088.

Certificate

If you have uploaded HTTPS certificates to S-Series, select it from the drop-down menu.

HTTP

The default port for HTTP is 80.

SSH

SSH port is used to access S-Series underlying configurations to debug the system. The default port is 8022. We recommend you disable SSH port if you do not need it.

FTP

With FTP service, you can connect to PBX via web browser. The default port is 21.

TFTP

To upload files to S-Series through TFTP, you need to enable this option.

IAX

The default port is 4569.

SIP UDP

The default port is 5060.

SIP TCP

The default port is 5060.

SIP TLS

The default port is 5061.

DHCP

Check the box Enable DHCP Server, S-Series will acts as a DHCP server. This feature is used when you do phone provisioning through DHCP mode.

S-Series VoIP PBX Administrator Guide

Figure 3-4 DHCP Server

 Gateway: enter the gateway IP address.  Subnet Mask: enter the subnet mask.  Preferred DNS Server: enter the preferred DNS server.  Alternate DNS Server: enter the alternate DNS server.  Allow IP Address: this sets the IP address that the DHCP server can assign to network devices.

Start IP address is on the left and end IP on the right.

 TFTP Server: this option is for Phone Provisioning feature. So IP phones can get configuration

file from this address. For Grandstream and Panasonic phones, enter the PBX’s IP address, for example: 192.168.5.150. For other IP phones, remember to specify the protocol, for example, tftp://192.168.5.150.

 NTP Server: the PBX can be a NTP server. By default, it is the PBX’s IP address.

AMI

The Asterisk Manager Interface (AMI) is a system monitoring and management interface provided by Asterisk. The 3rd party software can work with S-Series using AMI interface. The default port is

5080.

S-Series VoIP PBX Administrator Guide

Figure 3-5 AMI Settings

 Username: specify a name for the AMI user.  Password: specify a password for the user to connect to AMI.  Permitted IP/Subnet mask: configure permitted IP address and subnet mask that would be

allowed to authenticate as the AMI user. If you do not set this option, all IPs will be denied.

Certificate

S-Series supports TLS and HTTPS protocols. Before using these two protocols, you need to upload the relevant certificates to the system.

Click to upload a certificate.

Figure 3-6 Certificate

 Trusted Certificate: This certificate is a CA certificate. When selecting “TLS Verify Client” as

“Yes”, you should upload a CA. The relevant TLS client (i.e. IP phone) should also have this

certificate.

 PBX Certificate:

This certificate is server certificate. No matter selecting “TLS Verify Client” as ”Yes” or “NO”, you should upload this certificate to S-Series. If TLS client (i.e. IP phone) enables “TLS Verify server”, you should also upload the relevant CA certificate on IP phone.

Database Grant

Yeastar S-Series is using MySQL database. The 3rd party software can access MySQL via the Internet. Before that, you need to grant the authority to the database user. Go to Database Grant page, click

S-Series VoIP PBX Administrator Guide

to add a database user, specify the username and password.

Figure 3-7 Add Database Grant

 Username: configure the username which can be used by third party to access the database of

PBX.

 Password: configure the password which can be used by third party to access the database of

PBX.

 Permitted IP: enter the permitted IP address.

User Permission

The system has one default administrator account, which has the highest privileges. Here the administrator is referred as Super Admin. The system will automatically create user accounts when new extensions are created. By default, the extension users can log in the system and check their own settings and CDR. The Super Admin can grant more privileges for extension users. All the created users will be displayed on the User Permission page.

Figure 3-8 User Permission

 Super Admin has the highest privilege. The super administrator can access all pages on S-

Series Web and make all the configurations on the system. Username: admin Default Password: password

 Administrator is created by the Super Admin. The administrator has all the privileges but

cannot create new users for login.

 Custom User is created by the Super Admin. The Super Admin sets the privileges for those

users according to different situations.

S-Series VoIP PBX Administrator Guide

Add New User Permission

Log in the S-Series Web GUI with the Super Admin account, go to Settings > System > User Permission. Click to add a new User Permission. The following window prompts. Choose the

user and privilege type, then check the options to enable the privileges for the user.

Figure 3-9 Add New User Permission

Once created, the Super Admin can edit the users by clicking or delete the users by clicking .

User Portal

The extension user could log in S-Series Web GUI with the extension username and password. The extension user account is created automatically when an extension is created on the system.

 Username: extension number (i.e. 1000)  Default password: “pass” plus extension number (i.e. pass1000)

Below is an example of login page using extension number 1000.

S-Series VoIP PBX Administrator Guide

Figure 3-10 User Portal

Date & Time

Go to Settings > System > Date & Time to check the current time on the system. Here you can adjust time of the system (including time zone) to your local time.

Figure 3-11 Date & Time

 Time Zone: select your current time zone.  Daylight Saving Time: the option is disabled by default. Enable it when necessary.  Synchronize With NTP Server: if you choose this mode, the system will adjust its internal clock

to a central network server. Please note S-Series should be able to access the Internet if you

S-Series VoIP PBX Administrator Guide

choose this mode.

 NTP Server: enter a NTP server.

 Set Up Manually: if you choose this mode, you need to set the time manually.

 Date: choose the date.  Time: choose the time.

Set the system’s email to send voicemail to email, alert event emails, fax to email, email to SMS and SMS to email. Go to Settings > System > Email to configure the system email. Check the email settings parameters below.

Table 3-8 Email Settings

Option

Description

Email Address

Enter the email address.

Password

Enter the password.

Outgoing Mail Server (SMTP)

Enter SMTP server and port. Example:

smtp.sina.com:25

Incoming Mail Server (POP3)

Enter the POP3 server and port. Example:

pop.sina.com:110

Enable TLS

Use TLS to send secure message to server .If the email sending server needs to authenticate the sender, you need to select the checkbox.

Note: if you use Gmail or Exchange, you need enable this option.

After finishing the configuration, click to test the email. In the prompt, fill in an email address to send a test email to verify the Email settings.

Storage

Yeastar S-Series provides local storage (Flash) and supports external storage TF/SD card. Users could choose where to store the voicemails, CDR, recordings and logs.

Storage Devices

Go to Settings > System > Storage to configure the storage. All the local storage and external storage status shows on the page.

S-Series VoIP PBX Administrator Guide

Figure 3-12 Storage Devices

To format a external storage:

1. Click .

2. Click on the pop-up window to start formatting.

To add Network Drive:

The Network Drive feature is used to extend storage space. Before network drive can be properly configured, an SMB share folder accessible from Yeastar system must be set up on a Windows based machine. Once that has been set up, please follow the following instructions to configure network drive:

1. Choose a window-based computer that is always in service.

2. Create a folder.

3. Share this folder to Everyone.

4. Click and input the Net-Disk information in Yeastar S-Series:

Figure 3-13 Add Network Disk

 Name: give this network drive a name to help you identify it.  Host/IP: set the IP address where the recordings will be stored.

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 Share Name: the shared folder name where the recordings will be stored.  Access User Name: the User name used to log in the Network share. Leave this blank if it is not

required. In general, you use the administrator account on PC as a user name here.

 Access Password: the password used to log into the network share. Leave this blank if it is not

required.

5. If the configuration is correct, you can see the NETDISK status shown as below.

Figure 3-14 Network Drive Status

Storage Locations

When the storage devices are configured and ready to use, you can select where to store CDR, Recordings, Voicemail, one-touch recordings, logs.

Figure 3-15 Storage Locations

Auto Cleanup

Yeastar S-Series supports auto clean for CDR, logs, voicemails, one-touch recordings and recordings.

Table 3-9 Auto Cleanup Settings

CDR Auto Cleanup

Max Number of CDR

Set the maximum number of CDR that should be retained. The default is 100000. The old CDR will be deleted when the threshold is reached.

CDR Preservation Duration

Set the maximum number of days that CDR should be retained. The default is left blank.

Voicemail and One Touch Recording Auto Cleanup

Max Number of Files

Set the maximum number of voicemail and one touch recording files that should be retained. The default is 50. The old CDR will be deleted when the threshold is reached.

Files Preservation Duration

Set the maximum number of minutes that voicemails and one touch recordings should be retained. The default is left blank.

Recordings Auto Cleanup

Max Usage of Device

Set the maximum storage percentage the device is allowed to store. The default is 80%. The recordings will be deleted when the

S-Series VoIP PBX Administrator Guide

threshold is reached.

Recordings Preservation Duration

Set the maximum number of days that recording files should be retained. The default is left blank.

Logs Auto Cleanup

Logs Preservation Duration

Set the maximum number of days that logs should be retained.

“Logs Preservation Duration”. The default is 7.

This setting is for system log.

Max Number of Logs

Set the maximum number of logs that should be retained. The default is unlimited. The old logs will be deleted when the threshold is reached. This setting is for operation logs.

S-Series VoIP PBX Administrator Guide

Extensions

This chapter explains how to create and configure extensions on S-Series. Yeastar S-Series supports SIP, IAX and FXS extensions. An extension can be set to the 3 types and be registered to different devices. Go to PBX > Extensions page to configure the extensions.

 Add New Extension  Add Bulk Extensions  Search and Edit Extensions  Import and Export Extensions  Extension Group

Add New Extension

Click to add a new extension, you will see the pop-up window appear as below.

Figure 4-1 Add New Extension

Extension settings are divided to 4 categories:

 Basic  Feature  Advanced  Call Permission

Click on the tab to view or edit the relevant settings. Check the configuration parameters below.

Note: different settings would appear for different types of extension.

S-Series VoIP PBX Administrator Guide

 Basic Settings

Table 4-1 Extension Configuration Parameters – Basic

General

Type

Check the box to set the extension type. You can set the extension to multiple types.

 SIP  IAX  FXS: S2 or SO module should be installed on the device if you

want to create FXS extension.

Extension

The extension number that will be associated with this particular user or phone.

Caller ID

The Caller ID string that appears on outbound calls for this extension.

Registration Name

For extension registration validation.

Registration Password

The password for the user to register the SIP or IAX account. For example, 12t3f6.

Concurrent Registrations

Yeastar S-Series IP PBX supports SIP forking. SIP forking refers

to the process of “forking” a single SIP call to multiple SIP endpoints. The value of Concurrent Registrations limits how many SIP endpoints the extension can be registered.

User Information

Name

A character-based name for this user. For example, Bob Jones.

User Password

The password for this extension user to log in the system. For example, 12t3f6.

Email address of this extension user. The email will be used to recover password, receive forwarding voicemails, receive fax as an attachment, and receive event notifications.

Mobile Number

Mobile Number of this user. The number can receive forwarded calls and event notifications.

Prompt Language

The language of voice prompts. The default is the same with system language. If more language options are needed, please download it from "System Prompts" under "Voice Prompts".

 Features

Table 4-2 Extension Configuration Parameters – Features

Voicemail

Enable Voicemail

Check this box to enable voicemail for this extension.

Send Voicemail to Email

Check this box to send voicemail to the user's email address.

Note: to use this feature, "Email Settings" under "System" need to

be configured correctly.

S-Series VoIP PBX Administrator Guide

Voicemail Access PIN

Voicemail password used to access Voicemail system. This password can contain only numbers.

Call Forwarding

Always

Always redirect the call to the designated destination.

 Voicemail: redirect the caller to leave a voice message.  Extension: redirect the caller to another extension.  Users' Mobile Number: redirect the caller to the mobile number

filled in User Information.

 Custom Number: fill in the number manually and redirect the

caller to this number.

No Answer

Redirect the call to the designated destination when it is not answered.

When Busy

Redirect the call when the extension is busy.

Mobility Extension

Enable Mobility Extension

If you enable this setting, when the User's Mobile Number dial into the system, the phone will have the same user permission with the desktop extension. So the mobile number will be able to reach the other extension, dial out with the trunk, and play voicemail.

Mobility Extension

It is the same with the User's Mobile Number. A prefix matching the outbound route also needs to be filled in.

Ring Simultaneously

When the extension has an incoming call, it rings the mobile number simultaneously.

Monitor Settings

Allow Being Monitored

Check this option to allow this user to be monitored.

Monitor Mode

Decide how you will monitor another extension's current call.

 None: you will not be allowed to monitor other's call.  Extensive: all the following 3 modes will be available to use.  Listen: you can only listen to the call, but can't talk (default

feature code: *90).

 Whisper: you can talk to the extension you're monitoring

without being heard by the other party (default feature code: *91).

 Barge-in: you can talk to both parties (default feature code:

*92).

Other Settings

Ring Timeout

Customize the timeout in seconds. Phone will stop ringing over the time defined.

Max Call Duration

Select the maximum call duration in seconds for every call of this extension. If you wish to customize, enter the value in the text box directly. This option is valid only for outbound calls. If you choose “Follow System”, it would be equal to the “Max Call Duration” value in the “General” page.

S-Series VoIP PBX Administrator Guide

Call Waiting

Check this option if the extension should have Call Waiting

capability. If this option is checked, the “When busy” call forwarding

options will not be available. The call waiting function of IP phone has higher priority than MyPBX call waiting function.

DND

Don’t Disturb. When DND is enabled for an extension, the extension will not be available.

 Advanced Settings

Table 4-3 Extension Configuration Parameters – Advanced

VoIP Settings

NAT

This setting should be used when the system is using a public IP address, communicating 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.

Qualify

Check the box to send SIP OPTIONS regularly to the device to check if the device is still online.

Enable SRTP

Enable SRTP for voice encryption.

Check the box to allow registration of a remote extension.

Transport

Select the allowed transport.

DTMF Mode

Set the default mode for sending DTMF tones.

 RFC4733: DTMF will be carried in the RTP stream in different

RTP packets than the audio signal

 Info: DTMF will be carried in the SIP Info messages  Inband: DTMF will be carried in the audio signal  Auto: will use RFC4733 or Info automatically.

RFC4733 is the default mode.

IP Restriction

Enable IP Restriction

This option is used for IP access control. Check this option to enhance the VoIP security. Once enabled, only the IP address or IP section match the settings will be able to register this extension number.

Permitted IP/Subnet mask

Define the IP address or IP section which is allowed to register to the PBX. The input format should be IP address/Subnet mask.

Example:

 192.168.5.100/255.255.255.255 means only the device whose

IP address is 192.168.5.100 is allowed to register this extension number;

 192.168.5.0/255.255.255.0 means only the device whose IP

section is 192.168.5.XXX is allowed to register this extension number.

Analog Settings

S-Series VoIP PBX Administrator Guide

Min Flash Detection

Set the minimum amount of time, in milliseconds, that a hook flash must remain depressed in order for the system to consider it as a valid flash event. The default is 300 ms.

Max Flash Detection

Set the maximum amount of time, in milliseconds, that a hook flash must remain depressed in order for the system to consider it as a valid flash event. The default is 1000 ms.

Echo Cancellation

Enable or disable echo cancellation on the FXS port.

Rx Volume

The volume of the voice sent from the analog phone to the FXS port of PBX. Set the value from 5% to 100% or choose Custom to define the RX gain below.

Rx Gain

The gain of the voice sent from the analog phone to the FXS port of PBX. (Unit: db). The valid range is -30db to 6.0db.

Tx Volume

The volume of the voice sent from the FXS port of PBX to the analog phone. Set the value from 5% to 100% or choose Custom to define the TX gain below.

Tx Gain

The gain of the voice sent from the FXS port of PBX to the analog phone. (Unit: db) The valid range is -30db to 6.0db.

 Call Permission

Choose the outbound routes the user is allowed to use.

Figure 4-2 Call Permission

S-Series VoIP PBX Administrator Guide

Add Bulk Extensions

You can batch add SIP/IAX extensions on the system, which help you add a large amount of extensions quickly. Click to add extensions in bulk.

Figure 4-3 Add Bulk Extensions

Table 4-4 Bulk Add Extensions Configuration Parameters

General

Type

Choose the type for the extensions:

 SIP  IAX

Start Extension

Set the starting extension number of the batch of extensions to be added.

Create Number

The number of extensions to be created.

Decide which type of registration password will be used. There are 3 options.

 Random: generate a random password for each extension.  Fixed: use the text filled in as the password for all extensions.  Prefix + extension number: fill in a prefix and the password will

be the text plus the extension's number.

User Password

Decide which type of user password will be used. There are 3 options.

 Extension: use extension number as password for each

extension.

 Fixed: use the text filled in as the password for all extensions.  Prefix + extension number: fill in a prefix and the password will

be the text plus the extension’s number.

Concurrent Registrations

Set the max concurrent registrations for SIP extensions.

S-Series VoIP PBX Administrator Guide

Prompt Language

Set the language of voice prompt for extensions.

Search and Edit Extensions

All the extensions are listed on the extension page. Each extension has a checkbox for you to edit or delete in bulk. Also, you can edit or delete per extension by clicking or .

Figure 4-4 Extensions List

 Search Extension

You can search extensions by entering the extension number, name or type.

 Edit an Extension

Click to edit the desired extension.

 Delete an Extension

Click to delete the desired extension.

 Bulk Edit Extensions

Select the checkbox for the extensions, click to edit the extensions.

 Bulk Delete Extensions

Select the checkbox for the extensions, click to delete the extensions.

Importing and Exporting Extensions

Users could import and export extension configurations, which helps you manage extensions easily.

S-Series VoIP PBX Administrator Guide

To Import Extensions

1. Click , you will see a dialog window shown as below.

Figure 4-5 Import Extensions

2. Click Browse and select the file to start uploading. The file must be a .csv file. Check the sample

file below. You can export an extension file from the PBX and use it as a sample to start with.

Figure 4-6 Sample Extension File

S-Series VoIP PBX Administrator Guide

3. The sample csv file will result in the following extensions in the PBX.

Figure 4-7 Extension List

To Export Extensions

Select the checkbox of the extensions, click , the selected extensions would be exported to your local PC.

Figure 4-8 Export Extensions

S-Series VoIP PBX Administrator Guide

Extension Group

Extension Group feature allows you to assign and categorize extensions in different groups, which helps you to better manage the configurations in the system. For example, you can create Support and Sales groups, when configuring Outbound Route, you can select a extension group instead of each extension. This feature simplifies the configuration process.

Click to create an extension group.

Figure 4-9 Add Extension Group

S-Series VoIP PBX Administrator Guide

Trunks

Yeastar S-Series supports FXO trunk, BRI trunk, GSM/3G trunk, VoIP trunk and E1 trunk. In this chapter, we give a simplified guide of setting up trunks.

 FXO Trunk  BRI Trunk  GSM/3G Trunk  VoIP Trunk  E1/T1/J1 Trunk

FXO Trunk

FXO trunk is also known as PSTN trunk. The public switched telephone network (PSTN) is the network of the world's public circuit-switched telephone networks.

To extend FXO trunk on the system, you need to insert O2 or SO module to PBX. Go to Settings > PBX > Trunks to edit the FXO trunk. Before configuring a FXO trunk, please make sure that the analog line is connected to S-Series FXO port.

Click to edit the FXO trunk. Please check the FXO trunk configuration parameters below.

1) Basic Settings

Table 5-1 FXO Trunk Configuration Parameters – Basic

General

Trunk Name

Give this trunk a name to help you identify this trunk.

Rx Volume

Set the receiving volume of FXO port or choose Custom to define the RX gain below.

RxGain

The RX Gain for the receiving channel of FXO Port. The valid range is -30db to 12db.

Tx Volume

Set the transmitting volume of FXO port or choose Custom to define the TX gain below.

TxGain

The TX Gain for the transmitting channel of FXO Port. The valid range is -30db to 12db.

Enable SLA

If enabled, this trunk will not be available in routes or other channels.

Allow Barge

Whether to allow other SLA stations to join a call by pressing the SLA key.

Hold Access

Specify hold permission for the station.

 Open: other stations that share the same line could retrieve the call.  Private: the call can be retrieved only by the station that previously put the

call on hold, not by others sharing the same line.

2) Hangup Detection

Hangup detection settings help the system to detect if a call is hung up. If you find the PSTN call

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could not be disconnected, these settings need to be configured.

Table 5-2 Hangup Detection Configuration Parameters

Option

Description

Hangup Detection Method

Detect if a call is hung up with one of the following methods:  Busy Tone: listen for a busy tone to detect if the line got hung

up.

 Polarity Reversal: the call will be considered as “hang up” on a

polarity reversal.

Busy Count

Specify how many busy tones to wait for before hanging up. The default is 4. If you wish to customize, enter the value in the text box directly. Setting this too high might cause failure of busy detection.

Busy Pattern

Select the cadence of your busy signal. The default is None. If you wish to customize, enter the value in the text box directly. The input format should be "Sound,Silence". E.g. "500,500" means 500ms on, 500ms off.  If you choose None, the system will accept any regular sound-

silence pattern that repeats Busy Count times as a busy signal.

 If you specify Busy Pattern, the system will further check the

length of the tone and silence, which will further reduce the chance of a false positive disconnection.

Busy Interval

The busy detection interval. The default is 1. If you wish to customize, enter the value in the text box directly.

Frequency Detection

Decide whether to enable detecting the busy signal frequency or not.

Busy Frequency

If Frequency Detection is enabled, you must specify the local frequency. The default is 480,620. If you wish to customize, enter the value in the text box directly. Unit: Hz.

3) Answer Detection Type

Answer Detection will help the system to accurately bill your calls.

 None:  Polarity: choose this option if the FXO trunk could send polarity reversal signal after a call is

established.

4) Caller ID Settings

Caller ID Settings will help the system to detect Caller ID. If an incoming PSTN call does not display Caller ID, you need to confirm with your service provider if the line has enabled Caller ID feature. If this line does support Caller ID, configure these settings to solve this problem.

Table 5-3 Caller ID Configuration Parameters

Option

Description

Caller ID Detection

Whether to enable Caller ID detection.

S-Series VoIP PBX Administrator Guide

Caller ID Start

Define the start of a Caller ID signal. The options are:

 After Ring: detect Caller ID after first ring;  Before Ring: detect Caller ID before first ring;  After Polarity: detect Caller ID after polarity reversal;

The default is After Ring.

Caller ID Signaling

This option defines the type of caller ID signaling to use.

 Bell202  ETSI-V23  V23-Japan  DTMF

5) Other Settings

Table 5-4 Other Settings

Option

Description

Ring Detect Timeout

FXO (FXS devices) must have a timeout to determine if there was a hangup before the line is answered. This value can be used to configure how long it takes before the system considers a nonringing line with hangup activity. The default is 5000. If you wish to customize, enter the value in the text box directly. The valid range is 1000-8000.

Echo Cancellation

Whether to enable echo cancellation for this trunk.

Enable DNIS

Dialed Number Identification Service is a telephone service that enables a company to identify which telephone number was dialed. Users could configure DNIS to allow the IP phones to display which trunk is passing the call.

DNIS Name

A name for this DNIS, when a call reaches the selected trunk, the name will be displayed on the ringing phone.

BRI Trunk

Basic Rate Interface (BRI, 2B+D, 2B1D) is an Integrated Services Digital Network(ISDN) configuration intended primarily for use in subscriber lines similar to those that have long been used for plain old telephone service. The BRI configuration provides 2 bearer channels (B channels) at 64 kbit/s each and 1 data channel (D channel) at 16 kbit/s. The B channels are used for voice or user data, and the D channel is used for any combination of data, control/signalling, and X.25 packet networking.

To extend BRI trunk on the system, you need to insert B2 module to S-Series and connect the BRI port to the BRI provider with a RJ45-RJ11 cable.

Go to Settings > PBX > Trunks, click to edit the BRI trunk. Please check the BRI trunk configuration parameters below.

1) Basic Settings

Table 5-5 BRI Trunk Configuration Parameters – Basic

S-Series VoIP PBX Administrator Guide

General

Trunk Name

Give this trunk a name to help you identify this trunk.

Signaling

Specify the Signaling type according to the direction provided by your service provider.

Signaling Role

Specify whether this interface will act like the user or the network. The default is User.

Switch Type

Configure the switch type according to the direction provided by your service provider.

2) Advanced Settings

Table 5-6 BRI Trunk Configuration Parameters – Advanced

Advanced

Echo Cancellation

This option enables or disables echo cancellation. The default is checked.

Codec

Choose the codec for this trunk.

Facility-based ISDN Supplementary Services

Decide whether to enable transmission of facility-based ISDN supplementary services (such as caller name from CPE over facility) or not. The default is checked.

Overlap Dial

Define whether the system can dial this switch using overlap digits or not. If you need Direct Dial-in, then enable this option. The default is unchecked.

Reset Interval

This sets the time in seconds between restart of unused B channels. Set the internal to Never if you don't like the channel to restarts. The default is Never.

PRI Indication

Tells how PBX should indicate busy and congestion to the switch/user. The options are:  Inband: PBX plays indication tones without

answering; not available on all PRI/BRI subscription lines;

 Out-of-Band: PBX disconnects with busy/congestion

information code so the switch will play the indication tones to the caller.

The default is Out-of-Band.

Enable DNIS

DNIS Name

A name for this DNIS, when a call reaches the selected trunk, the name will be displayed on the ringing phone.

DID Number

This number is used to identify which line of the trunk is passing the call.

Hide Caller ID

Whether to hide caller ID or not.

S-Series VoIP PBX Administrator Guide

Dialplan

Calling Party Numbering Plan

Select the Calling Party Numbering Plan.

Calling Party Numbering Type

Select the Calling Party Numbering Type.

Called Party Numbering Plan

Select the Called Party Numbering Plan.

Called Party Numbering Type

Select the Called Party Numbering Type.

Presentation Indicator

The PI provides instructions on whether or not the provided calling line identity is allowed to be presented, or indicate that the number is not available.

Screen Indicator

The SI provides information on the source and the quality of the provided information.

ISDN Dialplan

ISDN/telephony numbering plan (Recommendation E.164)

International Prefix

Dialplan: '(Remote Dialplan:ISDN +) Remote Number Type: international'.

National Prefix

Dialplan: '(Remote Dialplan:ISDN +)Remote Number Type:national'.

Local Prefix

Dialplan: '(Remote Dialplan:ISDN +)Remote Number Type:subscriber'.

Private Prefix

Dialplan: 'Remote Dialplan:private + Remote Number Type: subscriber'.

Unknown Prefix

Dialplan: '(Remote Dialplan:ISDN +)Remote Number Type:unknown'.

3) DOD

DOD (Direct Outward Dialing) means the caller ID displayed when dialing out. Before configuring this, please make sure the provider supports this feature.  Global DOD

Configure Global direct outward dialing number. DOD (Direct Outward Dialing) is the caller ID displayed when dialing out. Before configuring this, please make sure the carrier supports this feature.

 Add one DOD with Multiple Extensions

Enter one DOD number and select multiple extensions.

S-Series VoIP PBX Administrator Guide

Figure 5-1 Add One DOD with Multiple Extensions

 Bind Consecutive DOD Numbers to Multiple Extensions

Enter the DOD number range and select the extensions.

Figure 5-2 Bind Consecutive DOD Numbers to Multiple Extensions

GSM/3G Trunk

Yeastar S-Series supports GSM/3G trunk. To extend the trunk, you need to install GSM/3G module to the S-Series and insert SIM card on the module. Click to edit the trunk. Please check the GSM/3G trunk configuration parameters below.

Table 5-7 GSM/3G Trunk Configuration Parameters

Option

Description

Trunk Name

Give this trunk a name to help you identify this trunk.

PIN Code

Enter the SIM card PIN code if the card has one.

Note: if you failed to enter your correct PIN code 3 times in succession, the SIM card will be permanently locked, which means you would need a new card.

Rx Volume

Set the receiving volume of GSM port or choose Custom to define the RX gain below.

RX Gain (db)

The RX Gain for the receiving channel of GSM Port. The valid range is 20db to 20db.

Tx Volume

Set the transmitting volume of GSM port or choose Custom to define the TX gain below.

TX Gain (db)

The TX Gain for the transmitting channel of GSM Port. The valid range is 20db to 20db.

Echo Cancellation

Whether to enable echo cancellation for the trunk.

Enable DNIS

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call.

DNIS Name

A name for this DNIS, when a call reaches the selected trunk, the name will be displayed on the ringing phone.

VoIP Trunk

Yeastar S-Series supports SIP and IAX protocols and provides 2 types of VoIP trunks:

 Register Trunk: registration based VoIP trunk. A Register Trunk requires S-Series to register with

the provider using an authentication name and password.

 Peer Trunk: IP based VoIP trunk. A Peer VoIP trunk does not require S-Series to register with the

provider. The IP address of S-Series needs to be configured with the provider, so that it knows where calls to your number should be routed.

Go to Settings > PBX > Trunks to add a VoIP trunk.

Please note that choosing different trunk protocol would have different settings.

1) Basic Settings

Table 5-8 SIP Register Trunk Configuration Parameters - Basic

SIP Register Trunk

Protocol

Set the trunk protocol “SIP”.

Trunk Type

Choose the trunk type “Register Trunk”.

Provider Name

Give this trunk a name to help you identify this trunk.

Transport

Set the transport method used by the trunk.

If Hostname/IP Address is the PBX’s Hostname and the port is 0 or

blank, NAPTR and SRV lookup will be executed to search for transport, port and server. If Hostname/IP Address is a legal IP address or a designated port, then UDP will be used.

Hostname/IP

Service provider’s hostname or IP address. The default SIP port is 5060.

Domain

VoIP provider’s server domain name. If the provider has no domain

name, fill in the IP address instead.

User Name

The username used to register to the trunk from the VoIP provider.

Password

The password to register to the trunk from the VoIP provider.

From User

All outgoing calls from the SIP trunk will use the From User (in this case the account name for SIP Registration) in From Header of the SIP Invite package. Keep this field blank if not needed.

Authentication Name

Used for SIP authentication. In most cases, it is the same with the username.

Enable Outbound Proxy

A proxy that receives requests from a client. Even though it may not be the server resolved by the Request-URI.

S-Series VoIP PBX Administrator Guide

Outbound Proxy Server

Configure the address of outbound proxy server. The address can be domain name or IP address.

Enable SLA

If enabled, this trunk will not be available in routes or other channels.

Allow Barge

Whether to allow other SLA stations to join a call by pressing the SLA key.

Hold Access

Specify hold permission for the station.

 Open: other stations that share the same line could retrieve the

call.

 Private: the call can be retrieved only by the station that

previously put the call on hold, not by others sharing the same line.

Table 5-9 SIP Peer Trunk Configuration Parameters - Basic

SIP Peer Trunk

Protocol

Set the trunk protocol as “SIP”.

Trunk Type

Choose the trunk type “Peer Trunk”.

Provider Name

Give this trunk a name to help you identify this trunk.

Transport

Set the transport method used by the trunk.

If Hostname/IP Address is the PBX’s Hostname and the port is 0 or

blank, NAPTR and SRV lookup will be executed to search for transport, port and server. If Hostname/IP Address is a legal IP address or a designated port, then UDP will be used.

Hostname/IP

Service provider’s hostname or IP address. The default SIP port is 5060.

Domain

VoIP provider’s server domain name. If the provider has no domain name, fill in the IP address instead.

Enable SLA

If enabled, this trunk will not be available in routes or other channels.

Allow Barge

Whether to allow other SLA stations to join a call by pressing the SLA key.

Hold Access

Specify hold permission for the station.

 Open: other stations that share the same line could retrieve the

call.

 Private: the call can be retrieved only by the station that

previously put the call on hold, not by others sharing the same line.

Table 5-10 IAX Register Trunk Configuration Parameters - Basic

IAX Register Trunk

Protocol

Set the trunk protocol “IAX”.

Trunk Type

Choose the trunk type “Register Trunk”.

Provider Name

Give this trunk a name to help you identify this trunk.

S-Series VoIP PBX Administrator Guide

Hostname/IP

Service provider’s hostname or IP address. The default IAX port is 4569.

User Name

The username used to register to the trunk from the VoIP provider.

Password

The password to register to the trunk from the VoIP provider.

Table 5-11 IAX Peer Trunk Configuration Parameters - Basic

IAX Peer Trunk

Protocol

Set the trunk protocol “IAX”.

Trunk Type

Choose the trunk type “Peer Trunk”.

Provider Name

Give this trunk a name to help you identify this trunk.

Hostname/IP

Service provider’s hostname or IP address. The default IAX port is 4569.

Domain

VoIP provider’s server domain name. If the provider has no domain

name, fill in the IP address instead.

2) Codec

Select codec for the VoIP trunk. Yeastar S-Series supports the codecs: a-law, u-law, GSM, iLBC, SPEEX, G722, G726, ADPCM, G729A, H261, H263, H263P, H264, MPEG4 and iLBC.

Figure 5-3 VoIP Trunk Codec

3) Advanced

Table 5-12 VoIP Trunk Configuration Parameters - Advanced

VoIP Settings

Qualify

Enable this to send SIP OPTIONS packet to SIP device to check if the device is up.

Enable SRTP

This option enables or disable SRTP (encrypted RTP) for the trunk.

T.38 Support

Whether to enable T.38 fax for the trunk.

DTMF Mode

Set the default mode for sending DTMF tones.

 RFC4733: DTMF will be carried in the RTP stream in different RTP

packets than the audio signal

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 Info: DTMF will be carried in the SIP Info messages  Inband: DTMF will be carried in the audio signal  Auto: will attempt to detect if the device supports RFC4733 DTMF.

If so, it will choose RFC4733; if not, it will choose Inband.

RFC4733 is the default mode.

Other Settings

Realm

Realm is a string to be displayed to users so they know which username and password to use. If you don’t know what to fill in, contact your service provider for further instructions.

Send Privacy ID

Check this checkbox to send privacy ID.

Enable DNIS

DID Number

This number is used to identify which line of the trunk is passing the call.

DNIS Name

A name for this DNIS, when a call reaches the selected trunk, the name will be displayed on the ringing phone.

4) DOD

DOD (Direct Outward Dialing) means the caller ID displayed when dialing out. Before configuring this, please make sure the provider supports this feature.  Global DOD

Configure Global direct outward dialing number. DOD (Direct Outward Dialing) is the caller ID displayed when dialing out. Before configuring this, please make sure the carrier supports this feature.

 Add One DOD with Multiple Extensions

Enter one DOD number and select multiple extensions.

Figure 5-4 Add One DOD with Multiple Extensions

 Bind Consecutive DOD Numbers to Multiple Extensions

Enter the DOD number range and select the extensions.

S-Series VoIP PBX Administrator Guide

Figure 5-5 Bind Consecutive DOD Numbers to Multiple Extensions

E1/T1/J1 Trunk

Yeastar S100 supports expanding up to 2 digital trunks, S300 supports expanding up to 3 digital trunks.

Go to Settings > PBX > Trunks to edit the digital trunk.

Please note that choosing different trunk signaling would have different settings.

1) Basic Settings

Table 5-13 PRI Trunk Configuration Parameters

PRI Signaling

Trunk Name

Give this trunk a name to help you identify this trunk.

Interface Type

Specify the interface type according to the trunk specification.

Signaling

Specify the Signaling type according to the direction provided by your service provider.

Framing

Choose the frame format for this trunk. When the Interface Type is E1, the options are:

· Enable CRC4

· Disable CRC4

CRC4 is a method of checking for errors in data transmitted on E1 trunk lines. When the Interface Type is T1 or J1, the options are:

· ESF

· D4

Line Code

Choose the line code for this trunk. When the interface Type is E1, the options are:

· HDB3

· AMI

When the Interface Type is T1 or J1, the options are:

· B8ZS

· AMI

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Codec

Choose the codec for this trunk.

Echo Cancellation

This option enables or disables echo cancellation. The default is checked.

D Channel

Set the channel used to carry control information and signaling information. When the Interface Type is E1, enter a channel number from 1 to 31. When the Interface Type is T1 or J1, enter a channel number from 1 to 24.

Switch Type

Configure the switch type according to the direction provided by your service provider.

Signaling Role

Specify whether this interface will act like the user or the network. The default is User.

Overlap Dial

Define whether the system can dial this switch using overlap digits or not. If you need Direct Dial-in, then enable this option. The default is Disable.

Table 5-14 MFC/R2 Trunk Configuration Parameters

MFC/R2 Signaling

Trunk Name

Give this trunk a name to help you identify this trunk.

Framing

Choose the frame format for this trunk. When the Interface Type is E1, the options are:

· Enable CRC4

· Disable CRC4

CRC4 is a method of checking for errors in data transmitted on E1 trunk lines. When the Interface Type is T1 or J1, the options are:

· ESF

· D4

Line Code

Choose the line code for this trunk. When the interface Type is E1, the options are:

· HDB3

· AMI

When the Interface Type is T1 or J1, the options are:

· B8ZS

· AMI

Echo Cancellation

This option enables or disables echo cancellation. The default is checked.

Variant

Set the MFC/R2 variant.

Category

Set the category of calling party.

MAX DNIS

Select max amount of DNIS to ask for.If you wish to customize, enter the value in the text box directly.

MAX ANI

Max amount of ANI to ask for.If you wish to customize, enter the value in the text box directly.

S-Series VoIP PBX Administrator Guide

Table 5-15 SS7 Trunk Configuration Parameters

SS7 Signaling

Trunk Name

Give this trunk a name to help you identify this trunk.

Framing

Choose the frame format for this trunk. When the Interface Type is E1, the options are:

· Enable CRC4

· Disable CRC4

CRC4 is a method of checking for errors in data transmitted on E1 trunk lines. When the Interface Type is T1 or J1, the options are:

· ESF

· D4

Line Code

Choose the line code for this trunk. When the interface Type is E1, the options are:

· HDB3

· AMI

When the Interface Type is T1 or J1, the options are:

· B8ZS

· AMI

Codec

Choose the codec for this trunk.

Echo Cancellation

This option enables or disables echo cancellation. The default is checked.

D Channel

Variant

Specify the SS7 Singalling variant. The options are:

· ITU: 14 bits

· ANSI: 24 bits

· China: 24 bits

Link set

Define SS7 linkset numbers.

Network Indicator

Specify the network indicator according to the network environment.

SLC

Specify the Signaling Link Code.

OPC

Specify the Originating Point Code. This is generally assigned by your carrier.

DPC

Specify the Destination Point Code. This is generally assigned by your carrier.

Table 5-16 E&M Trunk Configuration Parameters

E&M Signaling

S-Series VoIP PBX Administrator Guide

Trunk Name

Give this trunk a name to help you identify this trunk.

Interface Type

Specify the interface type according to the trunk specification.

Framing

Choose the frame format for this trunk. When the Interface Type is E1, the options are:

· Enable CRC4

· Disable CRC4

CRC4 is a method of checking for errors in data transmitted on E1 trunk lines. When the Interface Type is T1 or J1, the options are:

· ESF

· D4

Line Code

Choose the line code for this trunk. When the interface Type is E1, the options are:

· HDB3

· AMI

When the Interface Type is T1 or J1, the options are:

· B8ZS

· AMI

Codec

Choose the codec for this trunk.

Echo Cancellation

This option enables or disables echo cancellation. The default is checked.

2) Advanced

Table 5-17 PRI Trunk Configuration Parameters - Advanced

PRI Signaling

Facility-based ISDN Supplementary Services

Decide whether to enable transmission of facility-based ISDN supplementary services (such as caller name from CPE over facility) or not. The default is checked.

Reset Interval

This sets the time in seconds between restart of unused B channels. Set the interval to Never if you don't like the channel to restarts. The default is Never.

PRI Indication

Tells how PBX should indicate busy and congestion to the switch/user. The options are:

 Inband: PBX plays indication tones without answering;

not available on all PRI/BRI subscription lines;

 Out-of-Band: PBX disconnects with busy/congestion

information code so the switch will play the indication tones to the caller.

The default is Out-of-Band.

Enable DNIS

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DID Number

This number is used to identify which line of the trunk is passing the call.

DNIS Name

A name for this DNIS, when a call reaches the selected trunk, the name will be displayed on the ringing phone.

DialPlan

Calling Party Numbering Plan

Select the Calling Party Numbering Plan.

Calling Party Numbering Type

Select the Calling Party Numbering Type.

Called Party Numbering Plan

Select the Called Party Numbering Plan.

Called Party Numbering Type

Select the Called Party Numbering Type.

Presentation Indicator

The PI provides instructions on whether or not the provided calling line identity is allowed to be presented, or indicate that the number is not available.

Screen Indicator

The SI provides information on the source and the quality of the provided information.

ISDN Dialplan

ISDN/telephony numbering plan (Recommendation E.164)

International Prefix

Dialplan: '(Remote Dialplan:ISDN +) Remote Number Type: international'.

National Prefix

Dialplan: '(Remote Dialplan:ISDN +)Remote Number Type:national'.

Local Prefix

Dialplan: '(Remote Dialplan:ISDN +)Remote Number Type:subscriber'.

Private Prefix

Dialplan: 'Remote Dialplan:private + Remote Number Type: subscriber'.

Unknown Prefix

Dialplan: '(Remote Dialplan:ISDN +)Remote Number Type:unknown'.

Table 5-18 MFC/R2 Trunk Configuration Parameters - Advanced

MFC/R2 Signaling

Enable DNIS

DID Number

This number is used to identify which line of the trunk is passing the call.

DNIS Name

A name for this DNIS, when a call reaches the selected trunk, the name will be displayed on the ringing phone.

Forced Release

This option enables or disables forced release of channel. The default is unchecked.

Immediate Accept

Most variants of MFC/R2 offer a way to go directly to the call accepted state, by passing the use of group B and II tones. This option enables or disables the use of that feature for incoming calls. The default is unchecked.

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Double Answer

Block collect calls with double answer. This will cause that every answer signal is changed by answer -> clear back -> answer. The default is unchecked.

Charge Calls

Whether or not report to the other end "accept call with charge".

Allow Collect Calls

Specify whether to accept collect calls or not.

MF Back Timeout

MFC/R2 value in milliseconds for the MF timeout. The default is None.

Metering Pulse Timeout

MFC/R2 value in milliseconds for the metering pulse timeout. Enter -1 to use the default value.

DTMF Detection Timeout

Specify the DTMF Detection timeout in milliseconds.The default is 5000 ms.

Incoming DTMF Mode

Specify the incoming DTMF mode.

First Number of Get

Choose which number to get first.

Outgoing DTMF Mode

Specify the outgoing DTMF mode.

Table 5-19 SS7 Trunk Configuration Parameters - Advanced

SS7 Signaling

Enable DNIS

DID Number

This number is used to identify which line of the trunk is passing the call.

DNIS Name

A name for this DNIS, when a call reaches the selected trunk, the name will be displayed on the ringing phone.

Start CIC No.

Specify the Circuit Identification Code number of the first B channel of E1 line (SS7). Note: the suggested value is the multiples of 32 plus 1, for example: 1, 33, 65...

Calling Party Number Type

Calling Party Numbering Type

Called Party Number Type

3) DOD

DOD (Direct Outward Dialing) means the caller ID displayed when dialing out. Before configuring this, please make sure the provider supports this feature.  Global DOD

Configure Global direct outward dialing number. DOD (Direct Outward Dialing) is the caller ID displayed when dialing out. Before configuring this, please make sure the carrier supports this feature.

 Add One DOD with Multiple Extensions

Enter one DOD number and select multiple extensions.

S-Series VoIP PBX Administrator Guide

Figure 5-6 Add One DOD with Multiple Extensions

 Bind Consecutive DOD Numbers to Multiple Extensions

Enter the DOD number range and select the extensions.

Figure 5-7 Bind Consecutive DOD Numbers to Multiple Extensions

S-Series VoIP PBX Administrator Guide

Call Control

This chapter shows you how to control outgoing calls and incoming calls.

 Inbound Routes  Outbound Routes  Auto CLIP Routes  Time Conditions

Inbound Routes

When a call comes into S-Series from the outside, S-Series needs to know where to direct it. It can be directed to an extension, a ring group, a queue or a digital Receptionist (IVR) etc.

Go to Settings > PBX > Call Control > Inbound Routes to edit inbound routes. Please check the inbound route configuration parameters below.

1) Route Name

Give this inbound route a brief name to help you identify it.

2) DID Pattern

Match the DID Pattern in this field to pass incoming call through. Leave this blank to match calls with any or no DID info. You can use a pattern match to map a range of numbers. Only Peer to Peer Trunk, BRI Trunk, SIP Trunk need to configure this option. In patterns, the following characters have special meanings:

Table 6-1 DID Patterns Description

Patterns

Refers to any digit between 0 and 9

Refers to any digit between 1 and 9

Refers to any digit between 2 and 9

[###]

Refers to any digit in the brackets, example [123] is 1 or 2 or 3. Note that multiple numbers can be separated by commas and ranges of numbers can be specified with a dash ([1.3.6-8]) would match the numbers 1,3,6,7 and 8.

. (dot)

Wildcard. Match any number of anything.

Used to initiate call processing as soon as it can be determined that no other matches are possible.

If you want to route consecutive DID numbers to a range of consecutive extensions directly through SIP, SIP Peer to Peer, IAX Peer to Peer trunk, you need to enter the DID number range

(separate the first number and the last number by “-”), choose the Destination as Extension Range, and fill in the relevant extension numbers (separated by “-”).

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3) Caller ID Pattern

Define the Caller ID Number that is allowed to call in through this inbound route. Leave this field blank to match any or no CID info. You can also use patterns match to map a range of numbers. Press Enter to input multiple patterns.

4) Member Trunks

Select which trunks will be used in this route. To make a trunk a member of this route, please move it to the “Selected” box.

Figure 6-1 Member Trunks

5) Enable Time Condition

Decide if you want to route incoming calls based on Time Condition.

 If disabled, all calls will be routed to the Destination.  If enabled, you could route calls to different destinations at different time. Calls that do not

match the time periods will be routed to “Other Time” destination. The system will assign each Time Condition with a feature code, so you could use this code to force change the destination of a Time Condition and restore to its original destination.

Figure 6-2 Time Condition

6) Distinctive Ring Tone

The system supports mapping to custom ring tone files. For example, if you configure the distinctive ringing for custom ring tone to "Family", the ring tone will be played if the phone receives the incoming call.

7) Fax Detection

Decide if you want to enable Fax Detection.

 If disabled, the system will not detect fax tone nor will it send fax tone.

S-Series VoIP PBX Administrator Guide

 If enabled, the system will send the fax to Fax Destination if a fax tone is detected.

Fax Destination Sets the destination where to send the fax to. You can set it to:  Extension: send the fax to the designated extension. If it is a FXS extension, the fax will be

sent to the FXS fort (fax machine).

 Fax to Email: sent the fax as an email attachment to the designated email address, which

could be associated to an extension or a custom one.

Note: please make sure the sender email address is correctly configured in “System > Email”.

Outbound Routes

An outbound route works like a traffic cop giving directions to road users to use a predefined route to reach a predefined destination. Outbound routes are used to specify what numbers are allowed to go out a particular route. When a call is placed, the actual number dialed by the user is compared with the dial patterns in each route (from highest to lowest priority) until a match is found. If no match is found, the call fails. If the number dialed matches a pattern in more than one route, only the rules with the highest priority in the route are used.

Note:

 Yeastar S-Series compares the number with the pattern that you have defined in your route 1. If

matches, it will initiate the call using the selected trunks. If it does not, it will compare the number with the pattern you have defined in route 2 and so on. The outbound route which is in a higher position will be matched firstly.

 Adjust the outbound route sequence by clicking these buttons .

Go to Settings > PBX > Call Control > Outbound Routes to edit outbound routes. Please check the outbound route configuration parameters below.

1) Route Name

Give this outbound route a brief name to help you identify it.

2) Dial Patterns

Outbound calls that match this dial pattern will use this outbound route.

Table 6-2 Dial Patterns Description

Patterns

Refers to any digit between 0 and 9

Refers to any digit between 1 and 9

Refers to any digit between 2 and 9

[###]

S-Series VoIP PBX Administrator Guide

. (dot)

Wildcard. Match any number of anything.

Used to initiate call processing as soon as it can be determined that no other matches are possible.

Strip

Allow the users 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.

Prepend

Digits to prepend to a successful match. If the dialed number matches the patterns, then this will be prepended before sending to the trunks. 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 the calls are placed. When using analog trunks, a “w” character may also be prepended to provide a slight delay before dialing.

3) Member Trunks

Select which trunks will be used in this route.

Figure 6-3 Member Trunks

4) Member Extensions

Select extensions that will be permitted to use this outbound route.

S-Series VoIP PBX Administrator Guide

Figure 6-4 Member Extensions

5) Password

You can prompt users for a password before allowing calls to progress. The options are:

 None  PIN List: select a list of PIN  Password: enter a single password which will be needed when dialing through this outbound

route

6) Rrmemory Hunt

Round robin with memory, remembers which trunk was used last time, and then use the next available trunk to call out.

7) Time Condition

This defines the time conditions to use this outbound route.

Auto CLIP Routes

The system automatically stores information about outgoing calls to the AutoCLIP routing table. When a person calls back the call is routed directly to the original number. Go to Settings > PBX > Call Control > Auto CLIP Routes to configure Auto CLIP:

S-Series VoIP PBX Administrator Guide

Figure 6-5 Auto CLIP Route

 Record Keep Time: set the time duration for which records should be kept in the AutoCLIP

List. Default is 8 hours.

 Match Outgoing Trunk: if enabled, only the incoming call that came to the PBX through the

same trunk which made the call will be match against the AutoCLIP List.

 Member Trunks: choose the trunks, AutoCLIP Route will apply to the selected trunks.

Click View AutoCLIP List to view the records. In the AutoCLIP List you can see the record of the unconnected call.

Figure 6-6 Auto CLIP List

As the above figure shows, when the user (284288432) has a missed call and returns the call, he will be directly forwarded to extension 500 as shown in the AutoCLIP List.

SLA

Shared Line Appearance (SLA) feature helps users share SIP trunks and FXO trunks. It also helps

S-Series VoIP PBX Administrator Guide

monitor the status of the shared line. SLA feature works with BLF key on IP phones.

 When an incoming call is received, all the SLA stations are informed of it and may join it if the

shared line allows to barge in.

 When an outgoing call is made by one SLA station, all members shared with the same line are

informed about the call, and will be blocked from this line appearance until the line goes back to idle or the call is put on hold.

To use SLA, you need do the following:

 Enable SLA feature on a FXO trunk or VoIP trunk.  Create SLA Stations.  Configure BLF keys for the shared line on the stations' IP phones. The BLF key value is

“extension number_trunkname".

Go to Settings > PBX > Call Control > SLA, click to create SLA stations.

Figure 6-7 Add SLA Station

 Station Name: set a name for the SLA name.  Station: choose a SIP extension to monitor and use the SLA trunks.  Associated SLA Trunks: choose the SLA trunks.  Ring Timeout: set the ring timeout in seconds, phone will stop ringing after the time defined.  Ring Delay: set the delay time in seconds. Phone will delay ringing after the time defined. This

S-Series VoIP PBX Administrator Guide

time couldn’t be longer than “Ring Timeout”.

 Hold Access: specify hold permission for the station.

 Open: any station can place this trunk on hold and any other station is allowed to take it back

off of hold.

 Private: only the station that placed the trunk on hold is allowed to take it back off of hold.

Time Conditions

On Time Condition page, you can create time groups. A time group is a list of times against which incoming or outgoing calls are checked. The rules specify a time range, by the time, day of the week, day of the month, and month of the year. Time conditions can be assigned to an inbound route, which control the destination of a call based on the time. Time conditions can also be assigned to an outbound route in order to limit the use of that route.

Add Time Condition

Go to Settings > PBX > Call Control > Time Conditions, click to add time condition.

Figure 6-8 Add Time Condition

 Name: give this Time Condition a brief name to help you identify it.  Time: this is where you will define a time range. You can define multiple ranges in the same time

group by clicking .

 Days of Week: select a week day, month day, and/or month range in which you want this time

range to apply.

 Advanced Options: this option is disabled by default. If it is enabled, you need to set the month

and the day of the month. If it is disabled, it means that the time range defined above will apply to every day of the month, every month of the year.

Add a Holiday

After you have defined your office time conditions, you may need to create a holiday time groups. For example, you want to create a Holiday for Chinese National Day, from October 1st to October 5th.

Click to add a holiday.

S-Series VoIP PBX Administrator Guide

Figure 6-9 Add Holiday

Assigning Time Conditions to Inbound Routes

The created Time Conditions will become available for selection in the Inbound Routes.

Assigning Time Conditions to Outbound Routes

You can also assign Time Conditions to outbound routes, which may help you to control the route can be used. For example, you can limit the users to make outbound calls when your office is closed.

S-Series VoIP PBX Administrator Guide

Call Features

This chapter explains various call features on Yeastar S-Series.

 IVR  Ring Group  Queue  Conference  Pickup Group  Speed Dial  Callback  DISA  Blacklist/Whitelist  Pin List  Paging/Intercom  SMS

IVR

Like most organizations, where possible, we would like to route incoming calls an Auto Attendant. You can create one or more IVR (Auto Attendant) on S-Series to achieve it. When calls are routed to an IVR, S-Series will play a recording prompting them what options the callers can enter such as “Welcome to XX, press 1 for Sales and press 2 for Technical Support”.

Go to Settings > PBX > Call Features > IVR to configure IVR.

 Click to add a new IVR.

 Click to delete the selected IVR.

 Click to edit one IVR.

 Click to delete one IVR.

Please check the IVR configuration parameters below.

Table 7-1 IVR Configuration Parameters

Basic Settings

Number

Yeastar S-Series treats IVR as an extension; you can dial this extension number to reach the IVR from internal extensions.

Name

Give this IVR a brief name to help you identify it.

Prompt

The prompt that will be played when the caller reaches this IVR.

Prompt Repeat Count

The number of times that the selected IVR prompt will be played.

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Response Timeout

The number of seconds to wait for a digit input after prompt.

Digit Timeout

How long (in seconds) we wait for the caller to enter an option on their phone keypad before we consider it timed out and it follows the Timeout Destination as defined below.

Dial Extension

If this option is enabled, the callers can enter a user's extension number when entering the IVR to go direct to the users.

Dial Outbound Routes

Allow the caller to dial through outbound routes.

Keypress Events

Key Press Event 0 1 2 3 4 5 6 7 8 9 # * Timeout Invalid

Select the destination for each key pressing: digits 0-9, “#”, “*”, Timeout and Invalid. When the callers press the corresponding key, the call will be routed to:

 Extension  Voicemail  Ring Group  IVR  Conference Room  Queues  Faxes  Dial by Name  Hangup

Ring Group

A ring group helps you to ring a group of extensions in a variety of ring strategies. For example, you could define all the technical support guys' extensions in a ring group and ring the support guys one by one.

Go to Settings > PBX > Call Features > Ring Group to configure ring groups.

 Click to add a new ring group.

 Click to delete the selected ring groups.

 Click to edit one ring group.

 Click to delete one ring group.

Please check the ring group configuration parameters below.

S-Series VoIP PBX Administrator Guide

Table 7-2 Ring Group Configuration Parameters-General Settings

Option

Description

Number

The extension number dialed to reach this ring group.

Name

Give this ring group a brief name to help you identify it.

Ring Strategy

Select an appropriate ring strategy for this ring group.

 Ring All Simultaneously: ring all the available extensions simultaneously.  Ring Sequentially: ring each extension in the group one at a time.

Seconds to ring each member

Set the number of seconds to ring a single extension before moving to the next one.

Members

Choose the member of this ring group

Destination If No Answer

Choose the failover destination.

Queue

Queues are designed to receive calls in a call center. A queue is like a virtual waiting room, in which callers wait in line to talk with the available agent. Once the caller called in S-Series and reached the queue, he/she will hear hold music and prompts, while the queue sends out the call to the logged-in and available agents. A number of configuration options on the queue help you to control how the incoming calls are routed to the agents and what callers hear and do while waiting in the line.

Go to Settings > PBX > Call Features > Queue to configure queue.

 Click to add a new queue.

 Click to delete the selected queues.

 Click to edit one queue.

 Click to delete one queue.

Please check the queue configuration parameters below.

1) Basic Settings

Table 7-3 Queue Configuration Parameters - Basic Settings

Basic Settings

Number

Use this number to dial into the queue, or transfer callers to this number to put them into the queue.

Name

Give this queue a brief name to help you identify it.

Password

You can require agents to enter a password before they can login to this queue.

Ring Strategy

This option sets the Ringing Strategy for this Queue. The options are:  Ringing All: ring all available agents simultaneously until one answer.

S-Series VoIP PBX Administrator Guide

 Least Recent: ring the agent which was least recently called.  Fewest Calls: ring the agent with the fewest completed calls.  Random: ring a random agent.  Rememory: Round Robin with Memory, remembers where it left off in

the last ring pass.

 Linear: rings agents in the order specified in the configuration file.

Failover Destination

Set the failover destination.

Static Agents

This selection shows all users. Selecting a user here makes them a dynamic agent of the current queue. The dynamic agent is allowed to log in and log out the queue at any time.

 Dial "Queue number" + "*" to log in the queue.  Dial "Queue number" + "**" to log out the queue.

Agent Timeout

The number of seconds an agent's phone can ring before we consider it a timeout. If you wish to customize, enter the value in the text box directly.

Agent Announcement

Announcement played to the Agent prior to bridging in the caller.

Wrap-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 .If you wish to customize, enter the value in the text box directly. Input 0 for no delay.

Ring In Use

If set to “no”, unchecked, the queue will avoid sending calls to members whose device are known to be “in use”.

Retry

The number of seconds to wait before trying all the phones again. If you wish to customize, enter the value in the text box directly.

2) Caller Experience Settings

Table 7-4 Queue Configuration Parameters – Caller Experience Settings

Caller Settings

Music On Hold

Select the “Music on Hold” playlist for this Queue.

Caller Max Wait Time

Select the maximum number of seconds a caller can wait in a queue before being pulled out. If you wish to customize, enter the value in the text box directly. Input 0 for unlimited.

Leave When Empty

If enabled, callers already on hold will be forced out of a queue when no agents available.

Join Empty

If enabled, callers can join a queue that has no agents.

Join Announcement

Announcement played to callers once prior to joining the queue.

Caller Position Announcements

Announce Position

Announce position of caller in the queue.

Announce Hold Time

Enabling this option causes PBX to announce the hold time to the caller periodically based on the frequency timer. Either yes or no; hold time will be announced after one minute.

Frequency

How often to announce queue position and estimated hold time.

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Periodic Announcements

Prompt

Select a prompt file to play periodically.

Frequency

How often to play the periodic announcements.

Events

Once the events settings are configured, the callers are able to press the key to enter the destination you set. Usually, a prompt should be set on Periodic Announcements to guide the callers to press the key.

Conference

Conference Calls increase employee efficiency and productivity, and provide a more cost-effective way to hold meetings. Conference agents can dial * to access to the settings options and the admin can kick the last user out and can lock the conference room.

Go to Settings > PBX > Call Features > Conference to configure conferences.

 Click to add a new conference.

 Click to delete the selected conferences.

 Click to edit one conference.

 Click to delete one conference.

Please check the conference configuration parameters below.

Table 7-5 Conference Configuration Parameters

Options

Description

Number

Use this number to dial into the conference room.

Name

Give the conference a brief name to help you identify it.

Administrator

Admin can kick the users out and lock the conference. Also you can set none.

PIN#

You can require callers to enter a password before they can enter this conference. This setting is optional.

Join a Conference Room

Users on S-Series could dial the conference extension to join the conference room. If a password is set for the conference, users would be prompted to enter a PIN.

How to join the conference room if I am calling from outside (i.e. calling from my mobile phone)?

In this case, an inbound route for conferences should be set on S-Series. A trunk should be selected in the inbound route and destination should be set to a conference room. When the outside users dial in the trunk number, the call will be routed to the conference room.

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Manage the Conference

During the conference call, the users could manage the conference by pressing * key on their phones to access voice menu for conference room. Please check the options for the voice menu.

Table 7-6 Conference Voice Menu

Conference Administrator IVR Menu

Mute/ un-mute yourself.

Lock /unlock the conference.

Eject the last user.

Decrease the conference volume.

Increase the conference volume.

Decrease your volume.

Exit the IVR menu.

Increase your volume.

Conference Users IVR Menu

Mute/ un-mute yourself.

Decrease the conference volume.

Increase the conference volume.

Decrease your volume.

Exit the IVR menu.

Increase your volume.

Pickup Group

Call pickup allows one to answer someone else’s call. You can add pickup group. The default call

pickup for Group Call Pickup is *4. It allows you to pick up a call from a ringing phone which is in the same group as you.

Go to Settings > PBX > Call Features > Pickup Group to add pickup group.

 Click to add a new pickup group.

 Click to delete the selected pickup groups.

 Click to edit one pickup group.

 Click to delete one pickup group.

S-Series VoIP PBX Administrator Guide

Figure 7-1 Add Pickup Group

Speed Dial

Sometimes you may just need to call someone quickly without having to look up his/her phone number. You can by simply define a shortcut number. Speed Dial feature is available on Yeastar S-Series that allowing you to place a call by pressing a reduced number of keys.

1) Add Speed Dial

Click to add a speed dial.

Figure 7-2 Add Speed Dial

 Speed Dial Code: enter the speed dial code.  Phone Number: enter the number you want to call.

2) Import Speed Dial

Click , you will see a dialog window shown as below.

S-Series VoIP PBX Administrator Guide

Figure 7-3 Import Speed Dial Number

Click and select the file to start uploading. The file must be a .csv file. Check the sample file below. You can export a speed dial file from S-Series and use it as a sample to start

with.

Figure 7-4 Speed Dial File

The sample csv file will result in the following speed dial in Yeastar S-Series.

Figure 7-5 Speed Dial Codes

3) Export Speed Dial

Select the checkbox of the speed dial, click , the selected speed dial will be exported to

S-Series VoIP PBX Administrator Guide

your local PC.

Figure 7-6 Export Speed Dial

Callback

Callback feature allows callers to hang up and get called back to Yeastar S-Series Callback feature could reduce the cost for the users who work out of the office using their own mobile phones. Go to Settings > PBX > Call Features > Callback to configure Callback.

 Click to add a new callback.

 Click to delete the selected callbacks.

 Click to edit one callback.

 Click to delete one callback.

To use callback feature, you need to select callback as destination on the inbound route. Please check the callback configuration parameters below.

Note: you don’t need to configure “Strip” and “Prepend” options if the trunk supports call back with the caller ID directly.

Figure 7-7 Add Callback

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Table 7-7 Call Back Configuration Parameters

Option

Description

Name

Give this Callback a brief name to help you identify it.

Callback Through

Choose a trunk, the call will be called back through the selected trunk.

Delay Before Callback

Set the number of seconds before calling back a caller.

Strip

Defines how many digits will be stripped from the call in number before the callback is placed.

Prepend

Defines digits added before a callback number before the callback is placed.

Destination

The destination which the callback will direct the caller to.

DISA

DISA (Direct Inward System Access) allows someone calling in from outside Yeastar S-Series to obtain an “internal” system dial tone and make calls as if they were using one of the extensions of SSeries. 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 key (#). If the PIN number is correct, the user will hear dial tone on which a call may be placed. Please check the callback configuration parameters below.

Figure 7-8 Add DISA

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Table 7-8 DISA Configuration Parameters

Option

Description

Name

Give this DISA a brief name to help you identify it.

Password

The password for this DISA.

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 value is 10s.

Digit Timeout

The maximum amount of time permitted between each digit when the user is dialing an extension number. The default value is: 5s.

Member Outbound Routes

Defines the outbound routes that can be accessed from this DISA.

Blacklist/Whitelist

Blacklist is used to block an incoming/outgoing call. If the number of incoming or 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. Whitelist is used to allow incoming/outgoing numbers.

The system supports to block or allow 3 types of numbers:

 Inbound: the number would be disallowed or allowed to call in the system.  Outbound: users are disallowed or allowed to call the number out from the system.  Both: both inbound and outbound calls are disallowed or allowed.

1) Add Blacklist/Whitelist

Select Blacklist or Whitelist tag, click to add a number to Blacklist or Whitelist.

Figure 7-9 Add Blacklist

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 Name: give a name for the blacklist/whitelist.  Number: enter the numbers, one number per row.  Type: choose the type.

2) Import Blacklist/Whitelist

Click , you will see a dialog window shown as below.

Figure 7-10 Import Blacklist

Click and select the file to start uploading. The file must be a .csv file. Open the file with notepad, check the sample below. You can export a blacklist/whitelist file from S-Series

and use it as a sample to start with.

Figure 7-11 Blacklist/Whitelist File

The sample csv file will result in the following speed dial in Yeastar S-Series.

Figure 7-12 Blacklist/Whitelist

3) Export Blacklist/Whitelist

Select the checkbox of the blacklist/whitelist, click , the selected blacklist/whitelist will

S-Series VoIP PBX Administrator Guide

be exported to your local PC.

Pin List

PIN List is used to manage lists of PINs (numerical passwords) that can be used to access restricted features such as outbound routes. The PIN can also be presented in the CDR record.

Go to Settings > PBX > Call Features > Pin List and click to add Pin list.

Figure 7-13 Add PIN List

Linking a PIN List to Outbound Routes/DISA

After creating PIN lists, you can link the PIN lists to Outbound Routes or DISA. On outbound route/DISA edit page, you can select the PIN list from the Password drop-down menu.

Paging/Intercom

Intercom is a feature that allows you to make an announcement to one extension via a phone speaker.

The called party does not need to pick up the handset. It is can be achieved by pressing the feature code on your phone and it is a two-way audio call. The default Intercom feature code is *5. To make an announcement to a specific extension, you need to dial *5+ extension number on your phone. For example, make an announcement to extension 500, you need to dial *5500, then the extension 500 will be automatically picked up.

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. Paging is typically one way for announcements only, but you can set the paging group as a duplex mode to allow all users in the group to talk and be heard by all.

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Go to Settings > PBX > Call Features > Paging/Intercom, click to add a paging group.

Figure 7-14 Add Paging Group

 Number: the extension number dialed to reach this Paging Group.  Name: give this Paging Group a brief name to help you identify it.  Type: select the mode of paging group.

a) 1-Way Paging: typically one way for announcement only. b) 2-Way Paging: make paging duplex, allowing all users in the group to talk and be heard

by all.

 Member: select the members of the group.

SMS

Yeastar S-Series supports SMS to Email and Email to SMS features. To use these two features, you must do the following:

 Install GSM/3G module on the device.  Insert SIM card on the GSM/3G module.  Check the trunk status and make sure that the GSM/3G trunk is ready to be used.  Set an email address for the system (Settings > System > Email).

SMS to Email

SMS to Email is a feature that allows users’ email to receive the SMS of a GSM network. The SMS sent to the GSM/3G ports will be received first by application of Yeastar system and then forwarded to the pre-configured email address (the email set in Settings > System > Email). Thus, users can receive the SMS through email.

S-Series VoIP PBX Administrator Guide

Figure 7-15 Enable SMS to Email

Choose a GSM trunk and click , you will see the dialog appear as below. Click to add email

address then click .

Figure 7-16 Edit SMS To Email

When you send a SMS from your mobile to the GSM trunk number, the SMS message will be delivered to the email addresses.

Email to SMS

Email to SMS is a feature that allows users to send SMS to mobile phone number via email. When users would like to send a SMS, they just need to send an email to the Yeastar system's email address, with the destination mobile phone number as the email subject. The system will then receive the email and forward the email to the GSM/3G port, so that the email can be sent out through SMS to expected destinations.

Figure 7-17 Enable Email to SMS

S-Series VoIP PBX Administrator Guide

Sending Email to SMS, the Email subject format is as below: port:[port];num:[number];code:[code];

Note: for S100 and S300, you need point the GSM port is on which expansion board. For example, "port:2_1", means Expansion board 2 port 1 is GSM port.

1) Send Email to SMS without Access Code through default GSM/3G Port Email Subject: num:[number];

2) Send Email to SMS without Access Code through a Specific GSM/3G Port Email Subject: port:[port];num:[number];

3) Send Email to SMS with Access Code through Default GSM/3G Port Email Subject: port:[port];num:[number];code:[code];

4) Send Email to SMS with Access Code through a Specific GSM/3G Port Email Subject: port:[port];num:[number];code:[code];

S-Series VoIP PBX Administrator Guide

Voice Prompts

In this chapter, we introduce how to manage voice on Yeastar S-Series, including the following sections:

 Prompt Preference  System Prompt  Music on Hold  Custom Prompts

Prompt Preference

Select prompt files for the relevant options on this page.

Table 8-1 Prompt Preference Configuration Parameters

Option

Description

Music On Hold

The music to play when a call is being held.

Play Call Forwarding Prompt

If enabled, system will play a prompt before transferring the call. Otherwise, the call will be transferred directly without any prompt. It is enabled by default.

Music On Hold for Call Forwarding

This decides what to play when the caller is put on hold during call forwarding. The options are:  Music, which will be the same with the one selected in Music

on Hold.

 Ringing Tone The default is to play Music.

Invalid Phone Number Prompt

The prompt to play when the dialed phone number is invalid.

Busy Line Prompt

The prompt to play when the dialed phone number is busy.

Dial Failure Prompt

The prompt to play when a dial failed due to conjunction and lack of available trunks.

System Prompt

Yeastar S-Series ships with a US English prompt set by default. The system supports multiple languages. You could update the system prompt from the cloud server directly. Also, upload system prompt from local PC is supported. Go to Settings > PBX > Voice Prompt > System Prompt to update the system prompt.

Upload System Prompts

Click to select the system prompt file from local computer, then click to start uploading.

Figure 8-1 Upload System Prompts

S-Series VoIP PBX Administrator Guide

Download Online Prompt

Click , a dialog window appears as the following figure. All the available system prompts are listed on the window.

Figure 8-2 Download Online Prompt

Click to download the latest prompts. The new downloaded system prompt will be displayed once installed successfully. You can select the prompt to apply in the S-Series system or delete it.

Music on Hold

Music on hold (MOH) is the business practice of playing recorded music to fill the silence that

would be heard by callers who have been placed on hold. Users could configure Music on Hold Folder and upload music files to the system. The "default" Music on Hold Playlist includes 3 music files for users to use.

Go to Settings > PBX > Voice Prompts > Music on Hold.

1) Create New Playlist

Click to create a new playlist.

Figure 8-3 Add Playlist

S-Series VoIP PBX Administrator Guide

 Name: give this playlist a name to help you identify it.  Play Sort: select the playing order of the playlist.

2) Upload New Music

Figure 8-4 Upload New Music

Choose MOH Playlist from the drop-down menu. Click to select music file from your local computer, click to start uploading.

Custom Prompt

The default voice prompts and announcements in the system are suitable for almost every situation. However, you may want to use your own voice prompt to make it more meaningful and suitable for your case. In this case, you need to upload a custom prompt to the system or record a new prompt and apply it to the place you want to change. Go to Settings > PBX > Voice Prompts > Custom Prompts to record and upload custom prompts.

1) Upload Custom Prompt Click , the following dialog window appears. Click to choose a music file from

your computer. Click to start uploading.

Figure 8-5 Upload a Prompt

2) Record Custom Prompt

Click , the following dialog window shows. Specify the name and choose an extension to make the record.

S-Series VoIP PBX Administrator Guide

Figure 8-6 Record New Custom Prompt

Click , the selected extension will ring, pick up the call to start recording.

S-Series VoIP PBX Administrator Guide

General

This chapter explains general settings in the system, which can be applied globally to Yeastar SSeries.

 Preference  Feature Code  Voicemail  SIP  IAX

Preference

Table 9-1 Preference Configuration Parameters

Option

Description

Max Call Duration

Select the absolute maximum number of seconds permitted for a call. If you wish to customize, enter the value in the text box directly. Input 0 disables the timeout.

Attended Transfer Caller ID

The Caller ID that will be displayed on the recipient's phone. For example, Phone A (transferee) calls Phone B (transfer), and Phone B transfers the call to Phone C (recipient). If set to Transfer, the Caller ID displayed will be Phone B's number; if set to Transferee, Phone A's number will be displayed.

Virtual Ring Back Tone

Once enabled, when the caller calls out with cellular trunks, the caller will hear the virtual ring back tone generated by the system before the callee answers the call.

Distinctive Caller ID

When the incoming call is routed from Ring Group, Queue or IVR, the Caller ID would display where it comes from.

FXO Mode

Select a mode to set the On Hook Speed, Ringer Impedance, Ringer Threshold, Current Limiting, TIP/RING voltage, adjustment, Minimum Operational Loop Current, and AC Impedance as predefined for your country's analog line characteristics. The default setting is FCC for USA.

Tone Region

Select your country or nearest neighboring country to enable the default dial tone, busy tone, and ring tone for your region.

Extension Preferences

User Extensions

Specify the user extension range. The default range is 1000-5999.

Ring Group Extensions

Specify the Ring Group extension range. The default range is 6200-6299.

Paging Group Extensions

Specify the Paging Group extension range. The default range is 6300-6399.

Conference Extensions

Specify the Conference extension range.

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The default range is 6400-6499.

IVR Extensions

Specify the IVR extension range. The default range is 6500-6599.

Queue Extensions

Specify the Queue extension range. The default range is 6600-6699.

Feature Code

Feature Codes are used to enable and disable certain features available in the system. The S-Series local users can dial feature codes on their phones to use a particular feature. The default feature codes can be checked and changed via Settings > PBX > General > Feature Code.

Table 9-2 Feature Code

Feature Code

Feature Code Digits Timeout

The timeout to input next digit (in milliseconds). The default is

4000.

Recording

One Touch Record

The feature code that is used to start or stop call recording. The default feature code is *1.

Voicemail

Check Voicemail

The feature code that is used to check voicemail. The system will prompt you for password. The default feature code is *2.

Voicemail for Extension

You can leave a voicemail to other extensions by dialing feature

code on their phone or forward an incoming call to an extension’s

voicemail directly. The default feature code is #. For example, dial “#501” to leave a message for Ext. 501.

Voicemail Main Menu

The feature code that is used to access voicemail main menu. The default feature code is *02.

Transfer

Blind Transfer

Dial this feature code and an extension number to blind transfer the call. The default feature code is *03.

Attended Transfer

Dial this feature code and an extension number to transfer the call. Hang up after contacting the destination. The default feature code is *3.

Attended Transfer Timeout

The timeout to transfer a call, in seconds. The default is 15 seconds.

Call Pickup

This feature code allows you to answer another ringing phone that is in the same pickup group. The default feature code is *4.

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Extension Pickup

Dial this feature code and an extension number to pick up a call that is ringing at the extension. The default feature code is *04.

Intercom

Dial this feature code and an extension number to page that extension. The default feature code is *5.

Call Parking

Dial this feature code to put a call on hold and park the call at an extension number directed by the system. Any other phone can dial this extension number to resume the conversation. The default feature code is *6.

Directed Call Parking

Dial this feature code and an extension number to park the call at that extension. Any other phone can dial this extension number to resume the conversation. The default feature code is *6.

Note: if the directed extension number is occupied, the call parking will fail.

Parking Extension Range

A range of extensions where the call will be parked.

Parking Timeout

This defines the number of seconds that a call can be parked before it is recalled by an extension.

Call Forwarding

Reset to Defaults

Dial this feature code to restore call forwarding to the following default settings:

 Always Forward: disabled  Busy Forward to Voicemail: enabled  No Answer Forward to Voicemail: enabled  Do Not Disturb: disabled.

The default feature code is *70.

Enable Forward All Calls

Dial this feature code to forward all calls to voicemail or a designated number. For example: dial *71 to forward all calls to voicemail, and dial *71500 to forward all calls to number 500 (this number does cont include prefix, if you are required to dial with prefix, you need to configure it in Call Forwarding in Edit Extension window).

Disable Forward All Calls

Dial this feature code to disable forwarding of all calls. The default feature code is *071.

Enable Forward When Busy

Dial this feature code to forward calls to voicemail or a designated number when busy. For example: dial *72 to forward calls to voicemail when busy, and dial *72500 to forward all calls to number 500 when busy (this number does cont include prefix, if you are required to dial with prefix, you need to configure it in Call Forwarding in Edit Extension window). The default feature code is *72.

S-Series VoIP PBX Administrator Guide

Disable Forward When Busy

Dial this feature code to disable when busy call forwarding. The default feature code is *072.

Enable Forward No Answer

Dial this feature code to forward calls to voicemail or a designated number when no answer. For example: dial *73 to forward calls to voicemail when no answer, and dial *73500 to forward all calls to number 500 when no answer (this number does cont include prefix, if you are required to dial with prefix, you need to configure it in Call Forwarding in Edit Extension window). The default feature code is *73.

Disable Forward No Answer

Dial this feature code to disable no answer call forwarding. The default feature code is *073.

Call Monitor

Listen

Dial this feature code and the monitored extension number to initiate Listen monitoring. In this mode, the monitor can only listen to the call but can't talk. The default feature code is *90.

Note: to monitor an extension, you need to configure the Monitor Settings for this extension first.

Whisper

Dial this feature code and the monitored extension number to initiate Whisper monitoring. In this mode, the monitor can listen to and talk with the monitored extension without being heard by the other party. The default feature code is *91.

Note: to monitor an extension, you need to configure the Monitor Settings for this extension first.

Barge-in

Dial this feature code and the monitored extension number to initiate Barge-in monitoring. In this mode, the monitor can listen to and talk with both parties. The default feature code is *92.

Note: to monitor an extension, you need to configure the Monitor Settings for this extension first.

DND

Enable Do Not Disturb

Dial this feature code to put the extension in Do Not Disturb state. The default feature code is *74.

Disable Do Not Disturb

Dial this feature code to take the extension out of Do Not Disturb state. The default feature doe is *074.

Voicemail

The configurations of voicemail can be globally set up and managed on the Voicemail page. Go to

Settings > PBX > General > Voicemail, you can configure the Message Options, Greeting Options

and Playback Options.

S-Series VoIP PBX Administrator Guide

Table 9-3 Voicemail Configuration Parameters

Message Options

Max Messages per Folder

This sets the maximum number of messages that can be stored in a single folder of voicemail.

Max Message Time

This sets the maximum length of a single voicemail message (in seconds).

Min Message Time

This sets the minimum length of a single voicemail message (in seconds). Messages below this threshold will be automatically deleted.

Ask Caller to Dial 5

If this option is enabled, the caller will be prompted to press 5 before leaving a message.

Operator Breakout from Voicemail

If this option is set, the caller can jump out of the voicemail and go to the pre-configured destination by dialing 0.

Destination

This sets the breakout destination.

Greeting Options

Busy Prompt

Greeting played when the extension is busy.

Unavailable Prompt

Greeting played when the extension is unavailable.

Leave a Message Prompt

Greeting played when dial 5.

Playback Options

Announce Message Caller ID

If this option is enabled, the caller ID of the party that left the message will be announced before the voicemail message begins playing.

Announce Message Duration

If this option is enabled, the duration of the message in minutes will be announced before the voicemail message begins playing.

Announce Message Arrival Time

If this option is enabled, the arrival time of the message will be played back before the voicemail message begins playing.

Allow Users to Review Messages

Allow the callers to review their recorded messages before sending them to the voicemail box.

S-Series VoIP PBX Administrator Guide

Voicemail to Email Template

You can customize the Voicemail Email contents by clicking .

Figure 9-1 Voicemail To Email Template Settings

SIP

Go to Settings > PBX > General > SIP to configure SIP settings. It is wise to leave the default setting as provided on this page. However, for a few fields, you need to change them to suit your situation.

General

Table 9-4 General Settings

UDP Port

UDP Port used for SIP registrations. The default is 5060.

TCP Port

TCP Port used for SIP registrations. The default is 5060.

RTP Port

RTP Port for transmitting data. The From-port should start from 10000. From-port and To-port should have a difference value between 100 and 10000. The default is 10000-12000.

Local SIP Port

A random port in the port range will be used when sending packets to SIP server. The default range is 5062-5082.

Max Registration/Subscription Time

Maximum duration (in seconds) of incoming registrations and subscriptions. The default is 3600 seconds.

Min Registration/Subscription Time

Minimum duration (in seconds) of incoming registration and subscriptions. The default is 60 seconds.

Qualify Frequency

How often to send SIP OPTIONS packet to SIP device to check if the device is up. The default is 30 per second.

Outbound SIP Registrations

The number of registration attempts before giving up (0 for no limit).

S-Series VoIP PBX Administrator Guide

Default Incoming/ Outgoing Registration Time

Default duration (in seconds) of incoming/outgoing registration. The default is 120 seconds.

Note: the actual duration needs to minus 10 seconds from the value you filled in.

NAT

If your PBX is operating in a network connected to the internet through a single router, your PBX is behind NAT. The NAT device has to be instructed to forward the right inbound packets (from internet) to the PBX server. Usually you have to configure NAT settings when you want to register a remote extension to the PBX or when you need connect to the PBX via SIP trunk.

Yeastar S-Series supports 3 methods to configure NAT: STUN, External IP Address and External Host. You can select one method to configure NAT or disable NAT.

1) STUN

Figure 9-2 STUN

Table 9-5 STUN Configuration Parameters

Option

Description

STUN Address

Choose a STUN address in the drop-down list or customize with a STUN address and STUN port.

External Refresh Interval

If an external host has been supplied, you may specify how often the system will perform a DNS query on this host. This value is specified in seconds.

Local Network Identification

Used to identify the local network using a network number/subnet mask pair when the system is behind a NAT or firewall. Some examples are as follows:

“192.168.0.0/255.255.0.0”, “10.0.0.0/255.0.0.0”, and

“172.16.0.0/12”.

S-Series VoIP PBX Administrator Guide

NAT Mode

Global NAT configuration for the system. The options are as follows:  Yes: use NAT and ignore the 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 rport in headers.

2) External IP Address

Figure 9-3 NAT Settings – External IP Address

Table 9-6 External IP Address Configuration Parameters

Option

Description

External IP Address

The IP address that will be associated with outbound SIP messages if the system is in a NAT environment.

Local Network Identification

Used to identify the local network using a network number/subnet mask pair when the system is behind a NAT or firewall. Some examples are as follows:

“192.168.0.0/255.255.0.0”, “10.0.0.0/255.0.0.0”, and

“172.16.0.0/12”.

NAT Mode

Global NAT configuration for the system. The options are as follows:  Yes: use NAT and ignore the 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 rport in headers.

S-Series VoIP PBX Administrator Guide

3) External Host

Figure 9-4 NAT Settings – External Host

Table 9-7 External Host Configuration Parameters

Option

Description

External Host

Alternatively you can specify an external host, and the system will perform DNS queries periodically. This setting is only required when your external IP address is not static. It is recommended that a static public IP address be used with this system. Please contact your ISP for more information.

External Refresh Interval

If an external host has been supplied, you may specify how often the system will perform a DNS query on this host. This value is specified in seconds.

Local Network Identification

Used to identify the local network using a network number/subnet mask pair when the system is behind a NAT or firewall. Some examples are as follows:

“192.168.0.0/255.255.0.0”, “10.0.0.0/255.0.0.0”, and

“172.16.0.0/12”.

NAT Mode

Global NAT configuration for the system. The options are as follows:  Yes: use NAT and ignore the 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 rport in headers.

Codec

A codec is a compression or decompression algorithm that used in the transmission of voice packets over a network or the Internet. S-Series supports G711 a-law, u-law, GSM, H261, H263, H263P, H264, SPEEX, G722, G726, ADPCM, G719A, MPEG4 and iLBC.

Note:

If you would like to use G.729, please enter your license. The system have embedded the G729, you

S-Series VoIP PBX Administrator Guide

can test it directly without purchasing license. But for copyright protection, we suggest you to buy it after testing it successfully. After you buy the license from DIGIUM, you should enter G729 license at the "G729 License Key".

Figure 9-5 Codec Settings

TLS

Yeastar S-Series supports TLS protocol, to use TLS, you need enable TLS via Settings > PBX > General > SIP > TLS. Check the TLS configuration parameters below.

Table 9-8 TLS Configuration Parameters

Option

Description

Enable TLS

Check the checkbox to enable TLS.

TLS Port

TLS Port used for SIP registrations. The default is 5061.

Certificate

Choose the TLS certificates.

TLS Verify Server

If set to no, don't verify the servers certificate when acting as a client. If you don't have the server's CA certificate you can set this and it will connect without requiring TLS CA file. The default is no.

TLS Verify Client

If set to yes, verify certificate when acting as server. The default is no.

TLS Ignore Common Name

If set to yes, verify certificate when acting as server. The default is no.

TLS Client Method

Specify protocol for outbound client connections. The default is sslv2.

Session Timer

A periodic refreshing of a SIP session that allows both the user agent and proxy to determine if the SIP session is still active.

S-Series VoIP PBX Administrator Guide

Table 9-9 Session Timer Configuration Parameters

Option

Description

Session-timers

Choose the session timers mode on the system:

 No: do not include “timer” value in any field  Supported: include “timer” value in Supported header  Require: include “timer” value in Require header  Forced: include “timer” value in both "Supported" and "Required"

header.

The default is Supported.

Session-expires

The max refresh interval in seconds.

Session-minse

The min refresh interval in seconds, it must not be less than 90.

QOS

QoS (Quality of Service) is a major issue in VoIP implementations. The issue is how to guarantee that packet traffic for a voice or other media connection will not be delayed or dropped due interference from other lower priority traffic. When the network capacity is insufficient, QoS could provide priority to users by setting the value.

Figure 9-6 QOS

T.38

Figure 9-7 T.38

S-Series VoIP PBX Administrator Guide

 Re-invite SDP Not Add T.38 Attribute

If set to yes, SDP in re-invite packet will not add T.38 attributes.

 Error Correction

This sets the Error Correction Mode (ECM) for the Fax.

 T.38 Max BitRate

T38 Max Bit Rate.

Advanced

Table 9-10 SIP Advanced Settings

Option

Description

Allow RTP Re-invite

By default, the system will route media steams from SIP endpoints through itself. Enabling this option causes the system to attempt to negotiate the endpoints to route packets to each other directly, bypassing the system. It is not always possible for the system to negotiate endpoint-to-endpoint media routing.

Get Caller ID From

This decides the system will pull Caller ID header from which header field.

User Agent

This allows you to change the User-Agent field.

Get DID From

This decides the system will pull DID from which header field. If Remote-Party-ID is selected but the line does not support this, DID will be pulled from Invite header.

Send Remote Party ID

Whether to send the Remote-Party-ID in SIP header or not. The Default is no.

Send P Asserted Identify

Whether to send the P-Asserted-Identify in SIP header or not. The Default is no.

100rel

Check the option to enable 100rel.

Send Diversion ID

Whether to send the Diversion ID in SIP header or not. The Default is no.

Allow Guest

If enabled, it will allow the unauthorized INVITE coming into the PBX and the calls can be made. The default is no.

Jitter Buffer

Jitter is the variation in the time between packets arriving on a VoIP system. These variations can be caused by network congestion, timing drift or route changes. Jitter buffers are used to counter delay or latency, dropped packets, and jitter. They temporarily store arriving packets to minimize jitter and discard packets that arrive too late.

Yeastar S20, S50, S100, S300 Administrator's Manual

Specifications and Main Features

Frequently Asked Questions

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