Synway Syn_PBX80 U20V2, Syn_PBX80 U50V2, Syn_PBX80 U100, Syn_PBX100 U20V2, Syn_PBX100 U50V2 User Manual

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Contents

SAFETY NOTICE

.................................................................................................................................................

1 OVERVIEW

......................................................................................................................................................

1.1 BRIEF INTRODUCTION OFSYN_PBX SERIES 6

1.2 MAIN FEATURES 7

1.3 APPLICABLE MODULES 8

1.4 FEASIBLE MODULE COMBINATIONS 9

1.5 MECHANICAL DESIGN 10

1.5.1 U100 Front and Back View...................................................................................................... 10

1.5.2 LED Indication............................................................................................................................ 11

1.6 ENVIRONMENTAL REQUIREMENTS 12

1.7 PACKAGE CONTENTS 12

1.8 COMPATIBLE ENDPOINTS 12

2 GETTING STARTED

........................................................................................................................................

2.1 HARDWARE INSTALLATIONS 13

2.2 CONNECTSYN_PBX TO YOUR LAN 14

2.2.1 System Login..............................................................................................................................14

2.2.2 Configure Network Profiles...................................................................................................... 16

2.2.3 Module Configurations..............................................................................................................17

2.3 USER EXTENSIONS 18

2.3.1 New Extensions......................................................................................................................... 18

2.3.2 Other Extension Ranges.......................................................................................................... 19

2.4 IP EXTENSION REGISTRATION 19

2.4.1 Desktop IP phones.................................................................................................................... 19

2.4.2 Softphone on Windows PC...................................................................................................... 20

2.4.3 Softphone on Android phone, iPhone or iPad.......................................................................21

2.5 PHONE PROVISIONING 22

2.5.1 Phone Provisioning by PnP..................................................................................................... 22

2.5.2 Phone Provisioning by DHCP..................................................................................................24

2.6 ANALOG EXTENSIONS 24

2.7 EXTENSION STATUS 24

2.8 ADVANCED EXTENSION CONFIGURATIONS 25

2.8.1 Edit Properties of One Extension............................................................................................25

2.8.2 Search Extension...................................................................................................................... 27

2.8.3 Edit Properties of Multiple Extensions....................................................................................28

2.8.4 Upload/Download Extensions..................................................................................................29

3 SYN_PBX BASIC

............................................................................................................................................

3.1 TRUNKS 30

3.1.1 VoIP Trunk.................................................................................................................................. 30

3.1.2 FXO/GSM/WCDMA Trunk........................................................................................................ 32

3.1.3 E1/T1 Trunk................................................................................................................................ 34

3.1.4 BRI Trunk.................................................................................................................................... 35

3.2 OUTBOUND ROUTES 36

3.2.1 Dial Rules....................................................................................................................................36

3.2.2 Dial Plans....................................................................................................................................38

3.3 INBOUND CONTROL 39

3.3.1 Inbound Destinations................................................................................................................ 39

3.3.2 IVR............................................................................................................................................... 40

3.3.3 Ring Group................................................................................................................................. 42

3.3.4 Call Queue..................................................................................................................................43

3.3.5 Time Based Rules..................................................................................................................... 45

3.3.6 Office Closed Timing................................................................................................................. 46

3.3.7 Inbound Routes..........................................................................................................................47

4. SYN_PBX ADVANCED

..................................................................................................................................

4.1 GLOBAL SYN_PBX ADVANCED SETTINGS 49

4.1.1 General........................................................................................................................................49

4.1.2 Global Analog Settings............................................................................................................. 50

4.1.3 Global SIP Settings................................................................................................................... 52

4.1.4 Global IAX Settings................................................................................................................... 55

4.2 VIRTUAL FAX 56

4.2.1 Receive Fax................................................................................................................................56

4.2.2 Send Fax.....................................................................................................................................57

4.3 VOICEMAIL 58

4.3.1 General Voicemail Options...................................................................................................... 58

4.3.2 Playback Voicemail on the phone........................................................................................... 59

4.3.4 Voicemail to Email..................................................................................................................... 59

4.3.5 Playback Voicemail from Web GUI.........................................................................................61

4.4 CONFERENCE 62

4.4.1 Static Conference...................................................................................................................... 62

4.4.2 Dynamic Conference................................................................................................................ 63

4.5 MUSIC SETTINGS 64

4.6 DISA 65

4.7 FOLLOW ME 66

4.8 CALL FORWARD 67

4.8.1 Configure from the Web........................................................................................................... 67

4.8.2 Configure from the Phone........................................................................................................ 68

4.9 CALL TRANSFER 69

4.10 ONE NUMBER STATIONS 70

4.11 PAGING AND INTERCOM 71

4.12 WEB EXTENSIONS 72

4.13 PIN SETS 73

4.14 CALL RECORDING 74

4.14.1 Record All Calls....................................................................................................................... 74

4.14.2 One Touch Recording............................................................................................................. 75

4.15 SMART DID 75

4.16 CALLBACK 77

4.17 PHONE BOOK 78

4.18 LDAP SERVER 79

4.18.1 LDAP Server Settings............................................................................................................. 79

4.18.2 Synchronize Contacts with LDAP Server............................................................................ 79

4.18.3 LDAP Client Settings.............................................................................................................. 80

4.19 FEATURE CODES 81

5. NETWORK SETTINGS

...................................................................................................................................

5.1 NETWORK BASIC 84

5.1.1 IPv4 Settings.............................................................................................................................. 84

5.1.2 IPv6 Settings.............................................................................................................................. 86

5.1.3 VLAN Settings............................................................................................................................86

5.2 STATIC ROUTING 87

5.3 VPN 88

5.3.1 L2TP VPN................................................................................................................................... 88

5.3.2 PPTP VPN.................................................................................................................................. 90

5.3.3 OpenVPN.................................................................................................................................... 92

5.3.4 IPSec VPN..................................................................................................................................94

5.3.5 N2N VPN Client......................................................................................................................... 97

5.4 DHCP SERVER 97

5.4.1 DHCP Service............................................................................................................................ 97

5.4.2 DHCP Client List........................................................................................................................ 98

5.4.3 Static Mac................................................................................................................................... 98

5.5 DDNS 99

5.6 SNMPV2 100

5.7 TR069 100

5.8 TROUBLESHOOTING 101

5.8.1 Ping............................................................................................................................................101

5.8.2 Traceroute.................................................................................................................................102

5.8.3 TCPDUMP................................................................................................................................ 103

5.8.4 Channel Monitor...................................................................................................................... 103

6. REPORTS

....................................................................................................................................................

105

6.1 REGISTER STATUS 105

6.1.1 SIP User Status....................................................................................................................... 105

6.1.2 IAX2 User Status.....................................................................................................................105

6.1.3 SIP Trunk Status......................................................................................................................106

6.1.4 IAX2 Trunk Status................................................................................................................... 106

6.2 FAX LIST 107

6.3 RECORD LIST 107

6.3.1 Call Recording......................................................................................................................... 107

6.3.2 Conference............................................................................................................................... 108

6.3.3 One Touch Recording............................................................................................................. 109

6.3.4 Call Recording Playback........................................................................................................ 109

6.4 CALL LOGS 110

6.5 SYSTEM LOGS 110

7. SECURITY

...................................................................................................................................................

112

7.1 FIREWALL 112

7.2 SERVICE 114

7.3 FAIL2BAN 115

8. SYSTEM ADVANCED

..................................................................................................................................

117

8.1 TIME SETTINGS 117

8.1.1 NTP............................................................................................................................................ 117

8.1.2 Manual Time Set......................................................................................................................117

8.2 MODULE SETTINGS 118

8.2.1 E1 PRI Settings........................................................................................................................118

8.2.2 T1 PRI Settings........................................................................................................................119

8.2.3 BRI Settings............................................................................................................................. 120

8.2.4 MFC/R2 Settings..................................................................................................................... 122

8.2.5 SS7 Settings............................................................................................................................ 123

8.3 DATA STORAGE 123

8.3.1 USB Data Storage...................................................................................................................123

8.3.2 FTP Data Storage................................................................................................................... 124

8.4 MANAGEMENT 126

8.4.1 Administrator and Operator User Management................................................................. 126

8.4.2 Set System Voice Prompts.................................................................................................... 126

8.5 BACKUP 127

8.5.1 TAKE A BACKUP 127

8.5.2 UPLOAD BACKUP FILE 128

8.6 RESET & REBOOT 128

8.6.1 Reset......................................................................................................................................... 128

8.6.2 Reboot.......................................................................................................................................129

8.7 UPGRADE 130

8.7.1 Web Upgrade........................................................................................................................... 130

8.7.2 TFTP Upgrade......................................................................................................................... 130

Safety Notice

Please read the following safety notices before installing or using this IP PBX. They are crucial for

safe and reliable operation of the device. Failure to follow the instructions contained in this

document may result in damage to your PBX and void the manufacturer’s warranty.

1. Please use the external power supply which is included in the package. Other power

supplies may cause damage to the device, affect performance or induce noise.

2. Before using the external power supply in the package, please check your building power

voltage. Connecting to Inaccurate power voltage may cause fire and damage.

3. Please do not damage the power cord. If the power cord or plug is impaired, do not use it.

Connecting a damaged power cord may cause fire or electric shock.

4. Ensure the plug-socket combination is accessible even after the PBX is installed. In order to

service the PBX it will need to be disconnected from the power source.

5. Do not drop, knock or shake the device. Rough handling can break internal circuit boards.

6. Do not install the device in places where there is direct sunlight. Also do not place the

device on carpets or cushions. Doing so may cause the device to malfunction or cause a fire.

7. Avoid exposing the device to high temperature (above 40°C), low temperature (below -10°C)

or high humidity. Doing so could cause damage and will void the manufacturer warranty.

8. Avoid letting the device come in contact with water or any liquid which would damage the

device.

9. Do not attempt to open the device. Non-expert handling of the device could cause damage

and will immediately void the manufacturer warranty.

10. Consult your authorized dealer for assistance with any issues or questions you may have.

11. Do not use harsh chemicals, cleaning solvents, or strong detergents to clean the device.

12. Wipe the device with a soft cloth that has been slightly dampened in a mild soap and water

solution.

13. If you suspect your device has been struck by lightning, do not touch the device, power plug

or phone line. Call your authorized dealer for assistance to avoid the possibility of electric

shock.

14. Ensure the PBX is installed in a well-ventilated room to avoid overheating and damaging the

device.

15. Before you work on any equipment, be aware of any hazards involved with electrical

circuitry and be familiar with standard practices for preventing accidents if you are in a situation

that could cause bodily injury.

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V2.1.2

1 Overview

1.1 Brief Introduction of Syn_PBX Series

Syn_PBX Series IP Phone System is the most innovative solution for VoIP telecommunication in

the SMB (Small and Medium-sized Business) market. They provide not only traditional PBX

functionality such as automated attendant and voicemail, but also offer many

advanced telephony features, including remote extensions, remote office connection, IVR, call

recording, call detail records(CDR). All of these can serve to greatly enhance business operations

at reduced operational cost.

Syn_PBXSeries SYN_PBX is available in four model variants: U20V2, U50V2, U80 and U100.

This manual is dedicated for U80 and U100.

U80 and U100 share the same software and hardware architecture. The table below shows their

differences of system capacity.

Items

Syn_PBX80

Syn_PBX100

System

Capacity

Concurrent Calls

100

Extension Users

200

500

Voicemail and

Recording

1500 hrs (.gsm)

75000 hrs (.gsm)

150 hrs (.wav)

7500 hrs (.wav)

Conference Rooms

Hardware

Capacity

RAM

2GB DDR3L

4GB DDR3L

Storage

16GB EMMC

16GB EMMC + 500GB

Surveillance Hard Drive

U80 and U100 Contrast Table

Since U80 and U100 share the same software their configurations are the same, so in the manual

we take Syn_PBX100 as example to show you how to install, manage and use theSyn_PBX80

andSyn_PBX100 SYN_PBX systems.

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V2.1.2

1.2 Main Features

 BLF(Busy Lamp Field)

 Caller ID

 DND(Do Not Disturb)

 WebRTC

 Call Detail Records (20000 records)

 Call Center Queues (36)/ Callback

 Call Parking

 Call Forward

 Call Transfer

 Call Waiting

 Call Record

 Ring Group Record

 Call Queue Record

 Conference Bridge (36 Conferences)

 DISA (Direct Inward System Access)

 Paging and Intercom

 DID/DOD

 Smart DID

 Dial by Name

 Flexible Dial Plan

 Feature Codes

 SIP over IPv6

 One Number Stations

 Music On Hold

 Phonebook/LDAP(5000 contacts)

 Ring Group

 Speed Dial

 Skype for SIP

 SIP/IAX Extension Registration

 Static/DHCP/PPPoE Network Access

 System Backup

 T.38 Fax Pass-through

 USB Extended Storage (Scalable)

 Video Call

 Voicemail

 Virtual Fax

 Web-based Administration and configuration

 Extension User Portal

 Audio Codec: G.722/ G.711-Ulaw/ G.711-Alaw/ G.726/ G.729/ GSM/ SPEEX

 Video Codec: H.261/ H.263 / H.263+ /H.264

 VPN Server (L2TP/PPTP/OpenVPN/IPSec, up to 20 connections for VPN clients)

 VPN Client (L2TP/PPTP/OpenVPN/N2N/IPSec)

 DDNS(Dyndns.org/No-ip.com/zoneedit.com/ freedns.afraid.org/www.oray.com/ 3322.org)

 IP Phone Provisioning (Akuvox/Cisco/Escene/Fanvil/Grandstream/Htek/Yealink IP Phone)

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V2.1.2

1.3 Applicable Modules

4FXS Module

4FXO Module

2FXOS Module

2GSM Module

4GSM Module

2WCDMA

4WCDMA

E1/T1

4BRI

Notice:

1) Synway Module cards will only function in Syn_PBX IP PBX from Synway;

2) Module cards are packed separately but contained in the same package as theSyn_PBX system.

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V2.1.2

1.4 Feasible Module Combinations

For better performance please follow the feasible module combinations in the below table to

install you module cards. The combinations which have been marked as “No” are not

recommended and may cause module cards malfunction.

Slot 1

Slot 2

Feasibility

E1 Module

Vacant

4BRI Module

FXO/FXS Module

Yes

GSM/WCDMA Module

Yes

Vacant

E1 Module

Yes

4BRI Module

Yes

FXO/FXS Module

Yes

GSM/WCDMA Module

Yes

FXO/FXS Module

E1 Module

Yes

4BRI Module

Yes

FXO/FXS Module

Yes

GSM/WCDMA Module

Yes

GSM/WCDMA

E1 Module

Yes

4BRI Module

Yes

FXO/FXS Module

Yes

GSM/WCDMA Module

Yes

E1 Module

Yes

4BRI Module

FXO/FXS Module

GSM/WCDMA Module

4BRI Module

E1 Module

4BRI Module

FXO/FXS Module

GSM/WCDMA Module

Module Combination Table

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V2.1.2

1.5Mechanical Design

1.5.1 U100 Front and Back View

U100 Front View

No.

Name

Specification

LED Indicators

Indicate the system activaty and interface connection status.

U100 Back View

No.

Name

Specification

Power Switch

Switch the power on or off.

Power Cord

100~240V AC power.

PWR Button

Shutdown/Turn on the SYN_PBX system with power connected.

WAN Port

10/100/1000 Mbps.

LAN Port

10/100/1000 Mbps.

HDMI Port

For video output.

USB Port

For USB keyboard or USB storage.

Audio In/Out

For external paging.

Module Slots

For Synway Telephony Module Cards.

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V2.1.2

1.5.2 LED Indication

The LED indicators on the front panel indicate the interface connection and system activity status

of the Syn_PBX100

Identification

Indication

Status

Specification

PWR

Power States

Green

Power On

Off

Power Off

SYS

System States

Wink

System is Running

Off

System Booting or Failed

WAN/LAN

WAN/LAN Interface

States

Wink

Data Transmitting

Off

No Data Transmitting

1-4 (SLOT1/2)

Slot1 and Slot2

States

FXS

Green

Channel Loading Succeed

Wink

Channel Ringing

Off

Channel Loading Failure

FXO

Red

Channel Loading Succeed

Wink

Channel Ringing

Off

Channel Loading Failure

GSM/WCDMA

Red

Channel Loading Succeed

Wink

Channel Ringing

Off

Channel Loading Failure

E1/T1

(PRI/R2)

Red

Module Loading Succeed

Off

Module Loading Failure

L2/L3

Red/Off

CPE Signaling

Green/Off

NET Signaling

Off/Red

SS7 Signaling

Off/Green

R2 Signaling

Green

Connected (No Alarm)

Red

Disconnected (Alarm)

BRI

Red

TE Mode

Green

NT Mode

Off

Module Loading Failure

Syn_PBX100 LED Indication Table

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V2.1.2

1.6 Environmental Requirements

Operating Temperature: 0 °C ~40 °C

Storage Tempreture: -20 °C ~ 55 °C

Humidity: 5~95% Non-Condensing

1.7 Package Contents

Syn_PBX Main Case

Power Cable

Ethernet Cable

Quick Installation Guide

Warranty Card

Rack Mount Ear

Screws

1.8 Compatible Endpoints

 Any SIP compatible IP Phone (Desktop Phones and Soft Phones for Windows, Linux,

iOS and also Android platforms). Desktop phone examples include: CISCO,

Grandstream, Yealink, Polycom, Snom, Akuvox, Escene, Favil, HTek etc.Soft Phone

examples include 3CX, Linphone, X-Lite, Zoiper etc.

 IAX compatible endpoints

 Analog Phones and Fax Machines

 Web Extensions (WebRTC)

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V2.1.2

2 Getting Started

2.1 Hardware Installations

Before you can power on the Syn_PBX100, please read its Quick Installation Guide inside the packing box. There are some important notices about safety, environment and hardware installation prerequisites that you should be aware of. Please refer to the guide and properly install Syn_PBX100 in your computer room before turning it on and configure it. Below is what the cover of the Quick Installation Guide looks like.

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V2.1.2

2.2 Connect Syn_PBX to your LAN

2.2.1 System Login

Syn_PBX100 has been preconfigured with a static IP address of 192.168.1.100 on the devices

WAN port (192.168.10.100 on LAN port). If your network is configured with a different IP range to

the Syn_PBX system default address, then you will need to change the IP address to something

more appropriate before connecting to your local LAN.

Please connect your PC directly to the WAN interface of the SYN_PBX and change the network

profile of the PC to an IP address of 192.168.1.101 and Subnet mask of 255.255.255.0.

Now you can access the Web interface by inputting https://192.168.1.100:9999 into your

Internet browser address bar and pressing Enter.

You’ll now be presented with a Certificate Error notice as below, please click “Continue to this

website…”and you will be directed to the login page. Please ensure your IE browser version is at

least version 9 or you may not be able to access the web interface.

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V2.1.2

Enter the default username ‘admin’ and default password ‘admin’ to login in. After successful

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V2.1.2

within the notice to do this. To ensure the device is secure, the admin password must be complex

so please set a strong password that uses a combination of letters, numbers and also special

characters.

Notice:

1. SYN_PBX Web GUI supports the following 11 languages:

English, Chinese, Arabic, Persian, Portuguese, Italian, French, Spanish, Russian, Turkish and Thai.

You can select your native language or if this is not available then the most familiar one to login. We are

continuously adding more languages to meet the needs of our customers from all around the world.

2. Extension number can be used to login to the SYN_PBX Web GUI, for more details please check the Syn_PBX

user manual (Ext.User).

3. Operator user can login to the SYN_PBX Web GUI to monitor the system status and check call logs and faxes. By

default, operator user is disabled if you want to use operator user please enable it first. Please refer to chapter

8.3.1.

2.2.2 Configure Network Profiles

Navigate to Web Menu Network Settings-->Network->IPv4 Settings.

Syn_PBX WAN interface can be configured to operate in Static, DHCP or PPPoE mode. In the

majority of deployment scenario’s it is standard practice to configure the unit in Static mode.

DHCP and PPPoE will be described later in chapter 5.

To configure your Syn_PBX system in Static mode, you must assign an available static IP address

along with corresponding subnet mask, gateway and DNS to the WAN interface of the

Syn_PBX .For example, you could assign an IP address of 192.168.1.254, Subnet Mask:

255.255.255.0, Gateway: 192.168.1.1, DNS: 8.8.8.8.

After modifications are complete, please click the “Save” button to save the configuration. You

will now be presented with a dialog box asking you to reboot the system to make the changes

effective. Please reboot the system and once complete you can connect the SYN_PBX to your

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V2.1.2

local LAN switch.

2.2.3 Module Configurations

If you have installed FXO/FXS/GSM/WCDMA modules then you don’t have to configure module

settings. If you have E1/BRI modules installed then you’ll need to configure module settings for

the SYN_PBX system to load corresponding drivers and configure files.

To configure module settings please navigate to System->Module Settings page.

In most cases your E1/BRI modules will be installed on Slot2 and FXS/FXO/GSM/WCDMA will be

installed on Slot1. Your module settings will be similar as below examples.

Example 1: FXS/FXO/GSM/WCDMA module on Slot1, E1 module on Slot2.

The “Mode” parameter can set the module to work in E1 mode or T1 mode.

Example 2: FXS/FXO/GSM/WCDMA module on Slot1, BRI module on Slot2.

The E1/T1 and BRI settings should be given by your Telephone Company. Please configure the

E1/T1 and BRI parameters according to what they provided. More details of the parameters will

be introduced in chapter 8.2 Module Settings.

Notice:

1. Currently you can install two E1 modules onSyn_PBX80 and Syn_PBX100 systems but only one

BRI module on Slot2.

2. BRI module forSyn_PBX-V1 system cannot be used onSyn_PBX-V2, but new BRI modules

forSyn_PBX-V2 can be used onSyn_PBX-V1 system as well.

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V2.1.2

2.3 User extensions

Navigate to web menu Basic->Extensions.

This page lists all user extensions onSyn_PBX100 system. Here you can add/bulk add, delete/bulk

delete user extensions and also edit/bulk edit the user extension properties.

By default, 10 extension numbers within the range of 800 to 809 have been created for you to

use.

2.3.1 New Extensions

You can add further extensions one by one by clicking the “New User” button or bulk add

extensions by clicking “Batch Add” button and completing the popup shown below.

 Extension Start/Extension End: These two fields define the new extension range to be

generated.

 DialPlan: Select a dial plan for the new extensions.

 Password: A secure random password consisting of numbers, letters and special characters is

the recommended choice and can be selected by selecting the “Random” checkbox.

Alternatively, you can specify the same password for all new extensions. If you choose this

option then please ensure a secure password is set.

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V2.1.2

2.3.2 Other Extension Ranges

We have limited the user extension number range in the SYN_PBX to be between 800 and 899. If

you require more extensions or you want extensions in other number ranges then you need to

change the extension range before you can add new extensions.

Please navigate to web menu Advanced->Options->General.

In the “Extension Preferences” section you can change the user extension range.

In the above example, the user extension range has been changed to be between 100 to 599. If

you now go back to the extension page you’ll be able to add new extensions within this range.

2.4 IP Extension Registration

2.4.1 Desktop IP phones

The following example details how to register an IP phone on your Syn_PBX system.

Step 1:

Press the soft key “Status” beneath the phone screen, here you can see the IP Address of the IP

phone.

Step 2:

Open the IP phone web interface by entering the phone IP address into the web browser address

bar.

Step 3:

Default login credentials are username admin and password admin.

Step 4:

After successful login, navigate to the phone web menu VOIP->SIP, and register an extension

number as below example.

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V2.1.2

 Server Address: IP address of theSyn_PBX .

 Server Port: SIP service port number, by default this is 5060 and does not require changing.

 Authentication User: User extension number from the SYN_PBX user extension page.

 Authentication Password: The password of the extension.

 SIP User: The same as Authentication user.

 Display Name: Name of the extension user.

 Enable Registration: Only when enabled here will the phone register toSyn_PBX as an

extension.

2.4.2 Softphone on Windows PC

Softphones including 3CX, Bria, Zoiper and many other softphone APPs all work well

withSyn_PBX100. Below is an example of registering Zoiper to Syn_PBX system as an extension

from your Windows PC.

Step 1:

Download Zoiper from http://www.zoiper.com/.

Step 2:

Install and run Zoiper on your Windows device.

Step 3:

Click menu “Settings” and select “Create a new account” and select “SIP” protocol and click Next.

Step 4:

Complete the register credentials as in the example below:

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V2.1.2

Step 5:

Click Next to complete the registration process.

2.4.3 Softphone on Android phone, iPhone or iPad

The majority of softphones detailed previously in this section have mobile editions for both

Android and iOS platforms. You can download these apps and install them from your mobile

phone APP Store.

Below is an example of how to register Zoiper softphone to SYN_PBX as an extension from your

iPhone:

Step 1:

Run Zoiper on your iPhone and tap menu.

Step 2:

Tap menu.

Step 3:

Tap to create a new account.

Step 4:

You will be asked “Do you already have an account(username and password)?” tap “Yes” and

then tap “Manual configuration” to continue.

Step 5:

Tap to configure the account as in the below example:

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V2.1.2

Step 6:

After entering the register credentials, tap “Register” to register to SYN_PBX system as an

extension.

2.5 Phone Provisioning

If you plan to deploy a large number of IP phones, phone provisioning is a useful feature as it can

reduce the time and effort required to deploy p hone extensions. There are 2 methods to auto

provision your IP phones, DHCP and PnP.

2.5.1Phone Provisioning by PnP

Navigate to web menu Advanced->Phone Provisioning.

Here on this page you can see the term “PnP”, which refers to Plug and Play. By using this

technique you don’t have to undertake any configurations directly on the IP phones, but instead

only some minimized configurations on the SYN_PBX system. After this configuration is complete

you can plug the phones to your LAN and once they start up, they are ready for phone calls

through the SYN_PBX system.

Click on “PnP Settings” tab.

On this page, tick “Enable” to enable PnP feature.

 Interface: Select WAN or LAN depending on which interface you have connected the

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V2.1.2

SYN_PBX to your local LAN.

 Custom URL: Custom URL tells the IP phones where to obtain the configuration files for auto

provisioning. You should read your IP phone user manual to determine which kind of files it

requires for auto provisioning. Then you create/upload these files to a FTP/TFTP/HTTP server

for the phones to download. The URL can be IP address or domain name with subdirectory.

For example: http://192.168.1.2/phones/${MAC}.conf. With “Custom URL” configured, you

don’t have to add phones from the “Phone Settings” tab.

 Multicasting Address: IP phones which support PnP can use multicast discovery of SIP

Registrar. Multicast registrations are addressed to the well-known “all SIP servers” multicast

address “sip.mcast.net” (224.0.1.75 for IPv4).

 Port: SIP signaling port, default is 5060.

Notice:

Phone provisioning only works for IP phones that are in the same LAN where the Syn_PBX is deployed.

After enabling PnP feature, click on “Phone Settings” tab and click “New Phones” to generate the

configuration files for the phones to be added to the SYN_PBX system.

 Manufacturer: Manufacturer of the IP phone, currently, Syn_PBX supports phone

provisioning phones from the following manufacturers: Grandstream, Yealink, Escene,

AkuVox, Htek, Synway, Cisco, MOCET and Fanvil.

 Model: You must specify the exact model number of the phone, even if the phone is from

the same manufacturer. This is because different models require different configuration

files.

 MAC: Syn_PBX uses the MAC address of the phone to identify it on the local LAN as part of

the provisioning process and it essential that you enter the correct MAC address for your IP

Phone.

 Extension: The extension number selected here will be auto configured to the phone with

the MAC address given above.

 Label: Specify the user name of the phone.

Once you have added your new IP Phone(s) as described above, configuration files will be

generated in the background of the SYN_PBX system. You can now connect the phone(s) to your

LAN and once the phone(s) have booted up they will download configuration files from the

SYN_PBX system and complete auto configuration with the extension numbers you provided.

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2.5.2 Phone Provisioning by DHCP

If you want to auto provision your IP phones using DHCP, please make sure they support DHCP

option 66.

Please navigate to web menu: Network Settings->DHCP Server to enable DHCP service for the IP

phones first. Please refer to chapter 5.4.

Once DHCP is enabled you can add the phones in the same way as instructed above in Phone

Provisioning by PnP section, however, enabling PnP is not required in this scenario.

Notice:

If you are going to enable DHCP service on the Syn_PBX system, please ensure there is no other DHCP server in the

same LAN. If possible you can put the SYN_PBX and IP phones in a separate VLAN.

2.6 Analog Extensions

If yourSyn_PBX is equipped with an FXS port then you can configure an analog extension on your

SYN_PBX system. This can be an ordinary analog phone or it can be a fax machine for sending and

receiving faxes. The green LED indicates the RJ11 interface is FXS, you should connect the analog

phone/fax machine to the FXS port of the SYN_PBX.

Navigate to web menu: Basic->Extensions, click “New User” button to add an analog extension.

In the “Analog Phone” dropdown list, select an FXS port number for this new extension. This will

allow the analog phone/fax machine connected to this port to be assigned with this extension

number. The phone can now make and receive phone calls in the same manner SIP/IAX

extensions do.

2.7 Extension Status

You can check the status of all extensions configured on your Syn_PBX via the Operator page

“Operator” section.

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Here in this section, you can view real-time status of all extensions. Including idle(online), ringing,

in use and also on hold.

2.8 Advanced Extension Configurations

2.8.1 Edit Properties of One Extension

On the Basic->Extension page, you can click the “Edit” button to edit the properties of one

extension number.

Below are the explanations for the configuration options:

General

 SIP: Tick the checkbox to activate SIP protocol.

 IAX2: Tick the checkbox to activate IAX2 protocol.

 Name: Alias of this extension which can be the name of the extension user.

 Extension: Number of this user extension.

 Password: The password used for the phones to register.This can be set manually or can be

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generated by the SYN_PBX system. Auto generated password consists of numbers, letters

and special characters. If this is an analog extension then password is of no use as analog

phones are not required to register.

 Outbound CID: Choose a number to show to the external called party. This feature only

works with E1/T1, BRI and SIP trunks if the telco/ITSP(Internet Telephony Service Provider)

allows this number to be passed.

 DialPlan: Defines which type of numbers the extension can dial.

 Analog Phone: The FXS port number. An analog phone attached to this port will use this

extension number.

Voicemail

 Enable: Activate voicemail service for this extension.

 Password: Password for extension user to access the voicemail facilities.

 Delete VMail: Delete voice messages if the system has sent the message to user via email.

 Email: Email address of this extension user.

Other Options

 Web Manager: If enabled, users can use their extension number and voicemail password to

 Agent: If enabled, this user extension can be a call queue agent.

 Call Waiting: With this option enabled, busy extensions will hear the call-waiting tone, and

can use hook-flash to switch between callers. This option is only for analog extensions, for IP

extensions you have to configure this feature directly on the IP phones.

 Allow Been Spied: Enable this option to allow other extension users the ability to spy on the

phone calls of this extension by using feature codes.

 Pickup Group: Define a pickup group for this extension, extensions in the same pickup group

can help pick-up an incoming call on other ringing extensions in the same pickup group using

feature code *8. Available values are from 0 to 63.

 Mobility Extension: An external number can be specified here e.g. your mobile phone

number. If you now call the SYN_PBX using the mobile phone specified, you will hear a dial

tone and will have full access to the SYN_PBX system functionalities just as a standard

extension user does.

 Mobility Extension Number: When “Mobility Extension” as described above is enabled,

enter your external phone number here.

VoIP Settings

 NAT: Check this option if extension user or the phone is located behind a router.

 Transport: Choose UDP, TCP or TLS as the transport protocol for SIP signaling.

 SRTP: Secure Real-time Transport Protocol(SRTP) encrypts the RTP traffic to secure your VoIP

phone calls. Before enabling this option you need to ensure the end point can also support

SRTP.

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 DTMF Mode: Defines how the system detects DTMF tones, the default setting is rfc2833, it

can be changed if necessary.

 Permit IP: Defines which IP address or network address is allowed to register to this

extension number, other addresses will be unable to register. Addresses can be private or

public IP Addresses.

Video Options

 Video Call: Tick the checkbox to enable video call support. Supported video codecs are

H.261, H.263, H.263+, H.264.

Audio Codecs

Syn_PBXsupported audio codecs are G.711 (ulaw, alaw), G.722, G.726, G.729, GSM and Speex.

Enable the ones you require by moving the audio codecs to the “Allowed” column.

2.8.2 Search Extension

If there are too many extensions on the extensions page, it is difficult to locate a single extension

number to edit its properties, you can search by specifying the extension number and clicking on

“Search” button.

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2.8.3 Edit Properties of Multiple Extensions

Tick the checkboxes of the extensions you wish to edit, and click “Edit Selected” button and you

are able to edit the options as below:

If configured here, the selected extensions will have the same properties with the exception of

the extension numbers.

Notice:

Here you are configuring mutual parameters for the selected extensions, if you provide an IP address here in the

"Permit IP" field, then only the unique endpoint with this IP can register to these extensions. Only consider this if

these selected extensions are for an individual gateway or a remote office, otherwise please do not configure here

or please specify a network address.

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2.8.4 Upload/Download Extensions

The upload/download extensions feature can be used to backup or bulk add extensions of the

SYN_PBX system using text files. Supported file formats are CSV and TXT.

Click on the “Upload/Download Extensions” tab on Basic->Extensions page and you will see the

menu as below:

 Upload Extensions: Here you can upload .csv or .txt file to generate extensions.

 Download Extensions Template: Here you can download a template file in .csv or .txt format.

Inside there are examples which you can follow to add your desired new extensions in the

same format. Once complete, the new file can then be used to upload to Syn_PBX system to

generate new extensions.

 Download Extensions(.csv): Here you can download the existing extensions in the system for

backup. The downloaded CSV file can be used for extension list recovery.

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3 SYN_PBX Basic

3.1 Trunks

A trunk on an SYN_PBX system is essential for extensions to be able to make outbound phone

calls.Syn_PBX100 system (U80 and U100) support FXO, GSM, WCDMA, E1/T1, BRI and VoIP trunks

for outbound calls.

3.1.1 VoIP Trunk

Asterisk PBX can register as a SIP user agent to a SIP proxy (provider). If you have subscribed to a

VoIP service from an ITSP, then with the account details provided by them you can configure a

VoIP trunk on your Syn_PBX system for the user extensions to share and make outbound phone

calls.

Navigate to web menu Basic->Trunks. Click “New VoIP Trunk” button and complete the account

details provided to setup the trunk as in the example below.

 Description: A name for this trunk.

 Protocol: SIP or IAX2 protocol.

 Host: The SIP server domain or IP address.

 Maximum Channels: Maximum calls that can be made through this trunk at the same time,

0 means unlimited.

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 Prefix: The prefix number you enter here will be added in front of any number you dial via

this trunk. This feature is seldom required so please leave this field blank.

 Outbound Caller ID: The number you want to display to the called party.

 Without Authentication: If the service provider doesn’t require a username and password

for this account to register to their server then you can enable this option.

 Username: Username provided by VoIP Provider.

 Authuser: The optional authorization user for the SIP server

 Password: Password provided by VoIP Provider.

Advanced Options

 From Domain: Your service provider’s domain name.

 Insecure: Default value is “port, invite” ; “port”--Allow matching of peer by IP address

without matching port number; “invite”-- Do not require authentication of incoming

INVITEs.

 From User: fromuser=yourusername; Many SIP providers require this.

 Qualify(sec): Asterisk sends a SIP OPTIONS command regularly to check that the device is

still online. Default value is 2(sec).

 DID number: Self defined, and can be used to setup number DID.

 Transport: Default transport type for SIP messages.

 DTMF Mode: Used to inform the system how to detect the DTMF(Dual Tone Multi Frequency)

key press. Choices are inband, rfc2833, or info. By default we use RFC2833.

 NAT: With this option enabled, Asterisk may override the address/port information specified

in the SIP/SDP messages, and use the information (sender address) supplied by the network

stack instead. This feature is often required when there is a firewall located between the PBX

and the service provider.

 Context: Custom dial plan for this trunk, by default it uses the “default” dial plan. Configure

only if this trunk is for branch office integration, so calls coming from the other side can dial

out from this SYN_PBX trunk directly. DO NOT change unless you fully understand how this

feature works.

 Language: You can choose a desired language of the system voice prompts to play to the

incoming calls from this trunk. For example, if the call is not answered or the user is busy the

SYN_PBX system will notify the caller to leave a voice message in the language you set.

 Audio Codecs: Select the audio codec/codecs the provider can support.

 Video Codecs: If the ITSP supports video calls then you can enable compatible video codecs

here for video phone calls.

With the exception of configuration options related to your service provider and your account

details, please do not change the trunk advanced parameters if you are not familiar with them.

After the SIP trunk is successfully added you can see it listed here on this page.

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By clicking “Edit” you can modify the trunk settings and by clicking “Delete” you can remove this

trunk form the SYN_PBX system.

3.1.2 FXO/GSM/WCDMA Trunk

FXO Trunks

On the SYN_PBX front panel, red LED indicates the RJ11 interface is FXO. You should attach the

telephone wire from your telecom socket to the FXO ports. Once connected you should be able

to see the connection status on Operator page “FXO/FXS/GSM Ports” section.

To be able to make calls on your FXO interface you will first need to create a trunk(s). To create a

trunk you need to navigate to web menu Basic->Trunks->FXO/GSM Trunks.

Click “New FXO/GSM Trunk” button and you’ll see the available port numbers that can be used.

 Description: A name for this FXO trunk.

 Lines: Available FXO and GSM ports.

 Prefix: The prefix number you enter here will be added in front of any number you dial via

this trunk. This feature is seldom required so please leave this field blank.

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 Call Method: If in this trunk you have more than 1 FXO/GSM ports selected, then this

parameter defines how to use these ports for outbound phone calls.

 Busy Detection: Enable busy tone detection, it is also possible to specify how many busy

tones to wait for before hanging up.

 Busy Count: Specify how many busy tones to wait for before hanging up, configurable only if

Busy Detection is enabled.

 Input Volume: The volume of the incoming calls from FXO channel/channels.

 Output Volume: The volume of the outgoing calls from FXO channel/channels.

 Call Progress: If enabled, call progress attempts to determine answer, busy, and ringing on

phone lines. This feature is HIGHLY EXPERIMENTAL and can easily detect false answers and

therefore can prove unreliable.

 Progress Zone: Progress zone also affects the pattern used for busy detection, only effective

when Call Progress in turned on.

 Busy Pattern: If busy detection is enabled, it is also possible to specify the cadence of your

busy signal.

 Language: You can choose a desired language of the system voice prompts to play to the

incoming calls from this trunk. For example, if the call is not answered or the user is busy the

SYN_PBX system will notify the caller to leave a voice message in the language you set.

 Answer on Polarity Switch: When enabled, FXO (FXS signaled) ports watch for a polarity

reversal to mark when an outgoing call is answered by the remote party.

 Hangup on Polarity Switch: In certain countries, a polarity reversal is used to signal the

disconnection of a phone line. If the “hangup on polarity switch” option is selected, the call

will be considered "hung up" on a polarity reversal.

When creating a FXO trunk, if you are not competent with the advanced options then please do

not configure or change the default values.

GSM Trunk

If you have ordered GSM or WCDMA modules for yourSyn_PBX100 SYN_PBX system, the user

extensions will be able to make and receive phone calls from the mobile network.

You first have to insert SIM cards into the SIM slots of the GSM/WCDMA modules and then install

the modules to the SYN_PBX100 module slots. Antennas should be properly installed and placed

in open space for better signal reception. After completing the above, power on the SYN_PBX and

you’ll be able to configure GSM/WCDMA trunks in exactly the same way as you configure FXO

trunks.

GSM and WCDMA Specifications

Module

Working Frequencies

2GSM

GSM/GPRS 850/900/1800/1900MHz

4GSM

GSM/GPRS 850/900/1800/1900MHz

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2WCDMA

Quad-band: GSM850, EGSM 900, DCS 1800, PCS 1900.

SIM5320A: Dual-Band UMTS 850/1900MHz

SIM5320E: Dual-Band UMTS 900/2100MHz

SIM5320J: Dual-Band UMTS 850(800)/2100MHz

4WCDMA

3.1.3 E1/T1 Trunk

If you have E1 module installed you’ll get a new tab on Basic-Trunks page named E1/T1 Trunks.

Click on this tab and click on “New E1/T1 Trunk” you’ll be able to create a new E1/T1 trunk.

E1 connections have 32 channels in total, 30 channels are used as bearer channels (B channels)

and 2 channels are used as data channels (D channels). While T1 connections have 24 channels in

total, 23 channels are used as B channel and 1 channel is used as D channel.

In the above example, it’s an E1 connection so you have 30 available channels to be configured

for voice phone calls, if the module is configured to work as T1 then there will be 24 available

channels.

Below are the introductions of trunk configuration parameters.

 Description: A name to identify this E1 trunk.

 Channels: All available B channels of the E1 trunk.

 Prefix: The prefix number you enter here will be added in front of any number you dial via

this trunk. This feature is seldom required so please leave this field blank.

 Caller ID: The number you want to display to the called party.

 Call Method: Defines how to use these channels for outbound phone calls.

 Reset Interval: Sets the time in seconds between restart of unused B channels.

 Overlap Dial: Overlap dialing mode (sending overlap digits).

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 PRI Indication: Enable this to report Busy and Congestion on a PRI using out-of-band

notification.

 Language: Custom a system voice prompts language for the callers calling in from this trunk.

 Context: Rules of how to handle the inbound calls, default value is “Default” and it’s

recommended do not change it or you may not be able to receive inbound calls.

 Switch Type: Sets the type of PRI switch being used by the telephony provider.

 Auto Fax Detection: Automatically detect inbound faxes and send to specific destination.

3.1.4 BRI Trunk

If you have BRI module installed you’ll get a new tab on Basic->Trunks page named BRI Trunks.

Click on this tab and click on “New BRI Trunk” you are able to create a new BRI trunk.

The ISDN BRI configuration provides 2 bearer channels (B channels) and 1 data channel (D

channel). There are 2 channels each BRI port that can be used for voice phone calls. SYNWAY 4BRI

module has 4 BRI ports equipped, so in total there are 8 channels available for you to configure

BRI trunks.

Below are the introductions of trunk configuration parameters.

 Description: A name to identify this E1 trunk.

 Channels: All available B channels of the E1 trunk.

 Prefix: The prefix number you enter here will be added in front of any number you dial via

this trunk. This feature is seldom required so please leave this field blank.

 Caller ID: The number you want to display to the called party.

 Call Method: Defines how to use these channels for outbound phone calls.

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 Reset Interval: Sets the time in seconds between restart of unused B channels.

 Language: Custom a system voice prompts language for the callers calling in from this trunk.

 Context: Rules of how to handle the inbound calls, default value is “Default” and it’s

recommended do not change it or you may not be able to receive inbound calls.

 Switch Type: The ISDN switchtype must be set to match the switching equipment being used

by the telephony provider.

 Auto Fax Detection: Automatically detect inbound faxes and send to specific destination.

3.2 Outbound Routes

Outbound Routes allow you to define a set of dial rules that tell yourSyn_PBX which Trunks

(phone lines) to use when people dial external telephone numbers. A simple installation will

direct Syn_PBX to send all calls to a single trunk. However, a complex setup could have for

example an outbound route for emergency calls, another outbound route for local calls, another

for long distance calls, and perhaps even another for international calls.

With all of the above possibilities, you may have to configure several trunks on yourSyn_PBX

system and therefore you will need to configure several dial rules and maybe also several dial

plans.

3.2.1 Dial Rules

Navigate to web menu Basic->Outbound Routes->DialRules.

By default there are no existing dial rules configured in the SYN_PBX system. You need to click

“New DialRule” button to add a new dial rule.

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 Rule Name: A name for this dial rule.

 PIN Set: A collection of PIN codes for granting outbound phone calls. See chapter 4.13.

 Record in CDR: Record the PIN codes used for outbound phone calls along with the user

extension number and the dialed numbers in to the call logs.

 Call Duration Limit: Specify the maximum call time using this dial rule.

 Time Rule: Set a time condition when this dial rule can be used.

 Available Trunks: All existing trunks in the SYN_PBX system.

 Selected Trunks: Trunk/Trunks that can be used by this dial rule.

 Custom Pattern: Dial patterns act like a filter for matching numbers dialed with trunks. The

various patterns you can enter are similar to Asterisk's definition of them:

X — Refers to any digit between 0 and 9

N — Refers to any digit between 2 and 9

Z — Any digit that is not zero. (E.g. 1 to 9)

. — Wildcard. Match any number of anything. Must match *something*.

 Delete ____ digits prefix from the front and auto-add ________ digit before dialing: The first

blank allows you to strip some digit/digits before dialing out, here if required, you need to

complete the number of digits to delete. The second blank is to prepend some digit/digits

before dialing out, here you need to fill in the exact number of digits to be added in front of

the dialed number. For example a user dialing 912345678 using the dial rule example above,

the prefix 9 at the first digit will be removed, and 00 will be added, so eventually the number

called will actually be 0012345678.

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3.2.2 Dial Plans

Navigate to web menu Basic->Outbound Routes->DialPlans.

A default dial plan already exists in the SYN_PBX system. For most installations you just have to

click “Edit” button on the default dial plan “DialPlan1” and tick on all dial rules to enable them,

now extension users will be able to call any destinations using the trunk lines of the SYN_PBX

system.

Calling rules in the left column are for external calls and calling rules in the right column are for

internal calling. If you want to restrict some users from calling out through specific trunk lines or

you don’t want them to be able to call certain internal destinations, you can create a new dial

plan by clicking the “New DialPlan” button.

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In the new dial plan you should disable the rules you don’t want others to use and save. After this,

go to the extension configuration page and give the extension a different dial plan which ensures

the restrictions you made take effect.

3.3 Inbound Control

The Inbound Control section is where you define how Syn_PBX system handles incoming calls.

Typically, you determine the phone number that outside callers have called (DID Number) and

then indicate which extension, Ring Group, Voicemail, or other destination to which the call

should be directed.

3.3.1 Inbound Destinations

A call destination in Syn_PBX system might be an IVR menu that instructs the callers to press

certain digits to route their calls, a queue to wait for specific telephone services, a ring group to

call a number of user extensions, or virtually any other type of process to route the call in

whatever way is desired. A call may have several destinations throughout its lifespan.

Below is a list of call destinations available in Syn_PBX system:

 Extension

 Voicemail

 IVR

 Ring Group

 Paging Group

 Conference

 Call Queue

 DISA

 Time Rule

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 FAX

 Dial By Name

 Hangup

3.3.2 IVR

IVR, or interactive voice response, is responsible for the menus people hear and respond to when

they call up a company or business and hear the words for example: "press 1 for sales, press 2 for

marketing, press 0 to speak to the operator,".

IVR Prompts

To configure an IVR menu on Syn_PBX system you’ll first need to record your IVR prompts, these

IVR prompts will communicate to the callers the menu options that they have e.g. press one for

sales.

Navigate to web menu: Inbound Control->IVR Prompts

On this page you can delete the default voice prompts and click “New Voice” button to record a

new voice prompts from a designated extension.

Click “Record” button and the extension will ring, pick-up the extension and speak to record your

message. Once recording is complete your voice prompts will be listed on this page.

There is another way to add voice prompts to the system, click “Upload Voice Prompts” tab.

Here you can select a pre-recorded voice prompts file from your operating system to upload and

once complete your file will be listed on Voice Prompts page. Now you can use your file to setup

your personalized IVR menu.

IVR menu

Navigate to web menu: Inbound Control->IVR.

Click “New IVR” button to add an IVR menu.

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Let’s look at the above example where your IVR message says “Press 1 for sales, press 2 for

marketing, press 0 for operator”. If the caller is on the IVR menu, and after they hear the voice

prompts they press 1 then the sales ring group will ring, if 2 is pressed then the Marketing ring

group will ring, if 0 is pressed then will the IVR will ring the operator extension.

IVR Settings

 Name: Name for this IVR menu.

 Extension: Extension number for the IVR, by calling this number you can access the IVR

menu.

Welcome Message

 Please Select: Select a voice prompts for this IVR menu.

 Custom Prompts: Click this button to navigate to Inbound Control->IVR Prompts page for

new voice prompts.

 Repeat Loops: Define how many times to play the IVR menu to the caller.

 Timeout: Timeout for key pressing of each IVR loop.

 Dial other Extensions: If enabled, the caller can dial extension numbers directly when in the

IVR.

 Custom: By clicking “Custom” you can set a dial plan for this IVR menu and the callers on the

IVR will be able to dial other destinations that the dial plan allows.(Not recommended)

 Key Press Events: Define which destination to go by pressing a key on the phone keypad. If

undefined keys are pressed then they will be handled by the “i” parameter, “i” which means

invalid. And “t” stands for timeout, after all IVR loops are completed without the caller

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pressing any key then the incoming call will be handled by “t” parameter.

3.3.3 Ring Group

In a ring group, an incoming call will ring the phones of everyone in the group at the same time.

To configure a ring group please navigate to web menu: Inbound Control->Ring Groups.

Click “New Ring Group” button to add a ring group.

The extensions in the “Available Channels” column can be added to the ring group as a ring group

member.

 Name: Name for this ring group.

 Strategy: Defines how to ring the group members; selecting “RingAll” will ring all the

member extensions at the same time, selecting “Ring In Order” will ring the member

extensions one by one.

 Ring Group Members: The extensions selected to be the members of the ring group.

 Available Channels: All available extensions/channels can be added to the ring group.

 Label: Extensions can be members of multiple ring groups and therefore by giving each ring

group a different label, if an incoming call rings a ring group the label will be displayed on

the phone screen along with the caller ID. Therefore a ring group member will know which

ring group the call is coming from.

 Extension for this ring group: Reach the ring group member by calling this extension.

 Ring (each/all) for lasting time(sec): Ring duration of the group members.

 If not answered: Defines a destination to redirect incoming calls to if no one answers from

within the ring group.

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3.3.4 Call Queue

A call queue places incoming calls in line to be answered while extension users are busy with

other calls. The queued calls are distributed to the next available extension user in the order

received. Once a call queue has been created, it can be assigned to specific extensions and

configured to feature greetings, messages, and hold music.

To configure a call queue please navigate to web menu Inbound Control->Call Queue.

There are 3 existing call queues pre-configured and all you have to do is click “Edit” button to

configure them. If you require more call queues then click “New Call Queue” to add more.

Here we can see in the “Agents” field there are no available agents to be assigned to the call

queues. Click “click here” and you’ll be redirected to the extension page to determine which

extensions will be employed as call queue agents.

Tick the checkbox of the extension numbers which will be employed as call queue agents, then

click “Edit Selected” button and tick the “Agent” option in the “Other Options” section.

Save and go back to Inbound Control->Call Queues page again and now you will be able to

configure the existing call queues and add new call queues with available agents.

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 Queue Number: Define an extension number to identify the queue.

 Label: Define the label for the queue. A user can be an agent of multiple queues, by giving a

label for the call queue, if an incoming call is distributed to an agent the label will be

displayed on the phone screen along with the caller ID. So a call queue agent knows which

call queue the call is coming from.

Ring Strategy

 RingAll: Ring all available agents until one answers(default)

 RoundRobin: Starting with the first agent, ring the extension of each agent in turn until the

call is answered.

 LeastRecent: Ring the extension of the Agent who has least recently received a call

 FewestCalls: Ring the extension of the Agent who has taken the fewest number of calls.

 Random: Ring the extension of a random Agent.

 RRmemory: RoundRobin with Memory, like RoundRobin above, except instead of the next

call starting with the first agent, the system remembers which extension was last called and

begins the round robin with the next agent .

 Agent: Check each agent that you want to be a member of this specific Call Center Queue.

 Agent TimeOut(sec): Specify the number of seconds to ring an agent’s extension before

sending the call to the next Agent (based on Ring Strategy).

 Auto Pause: If an Agent’s extension rings and the Agent fails to answer the call,

automatically pause that agent to stop them receiving further calls from the queue.

 Wrap-Up-Time(sec): This is the amount of time in seconds that an agent has to complete

work on a call after which the call is disconnected. (Default is 0, which means no wrap-up

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time.)

 Max Wait Time(sec): Calls that have been waiting in the queue for this number of seconds

will be sent to the “If not answered” destination.

 Max Callers: Max number of callers who are allowed to wait in the queue. (Default is 0,

which means unlimited.) when the maximum number of callers in the queue is reached,

subsequent callers will be sent to the “If not answered” destination.

 Join Empty: Allow callers to enter the Queue when no Agents are available. If this option is

not defined, callers will not be able to enter Queues without available agents - callers will be

sent to the “If not answered” destination.

 Leave When Empty: If this option is selected and calls are still in the queue when the last

agent logs out, the remaining callers in the Queue will be transferred to the ”If not answered”

destination. This option cannot be used with Join Empty simultaneously.

 Auto Fill: Callers will be distributed to Agents automatically.

 Report Hold Time: Report the hold time of the next caller for Agent when the Agent is

answering the call.

 Frequency(sec): Repeat frequency to announce the hold time for callers in the Queue.(“0”

means no announcement).

 Announce Hold Time: Announce the hold time. Announce (yes), do not announce (no) or

announce once (once), There will be no announcement when the hold time is less than 1

minute.

 Repeat Frequency(sec): Interval time to play the voice menu for callers.(“0” means do not

play).

 Announcement Prompt: Select an IVR prompt to be used as the Announcements Prompt.

3.3.5 Time Based Rules

Many businesses have fixed working hours where they know for example that they are only open

Monday to Friday between 9am and 5pm and will be closed for business at all other times. Time

conditions inSyn_PBX allow you to control what happens to inbound calls both during and

outside normal business hours.

Navigate to web menu: Inbound Control->Time Based Rules.

Click “New Time Rule” to add a time condition for the system:

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Here you configure the time and date of your business hours. If you receive a call where the time

matches then the inbound call will be directed to “office-hours” IVR menu. If the time does not

match then the inbound call will be directed to “closed time” IVR menu.

3.3.6 Office Closed Timing

Office closed timing is an extending of time based rules, you can manually activate office closed

timing by feature code. This feature allows much more flexible time conditions to be temporarily

applied for the offices which may have some unscheduled businesses and activities off the time

table of the time based rule/rules.

For example, the office opens in the morning but there’s an event in the afternoon and by then

nobody will be able to answer phone calls. You can direct the inbound calls to an extension’s

voicemail or the closed time IVR.

 Enable Office Closed Timing: By dialing the feature code on the phone, you can activate

office closed timing. (Default is *81)

 Disable Office Closed Timing: By dialing the feature code on the phone, you can deactivate

office closed timing. (Default is *081)

 Destination: The destination of the inbound calls while office closed timing is activated. It

needs to be pre-configured before you can use this feature.

 Save: Save the settings of office closed timing.

 Cancel: Cancel the settings.

 Status: Status of office closed timing, “Enabled” or “Disabled”.

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3.3.7 Inbound Routes

General

For both FXO channels and VoIP channels, you can define default inbound destinations. If you

don’t want the inbound calls to always go to an IVR menu, ring group or extension, then you can

use a time rule to handle the inbound calls.

Port DIDs

If some of the FXO/GSM/WCDMA ports are dedicated to a specific calling service and you want

them handling differently to your generic service then you can configure “Port DIDs” here.

For the above example, all inbound calls from FXO port 1 will be directed to extension number

401. General inbound control will still work with other ports which have not been configured

with port DIDs.

Number DIDs

Number DID is for inbound control of VoIP/E1/T1/BRItrunks but not FXO/GSM/WCDMAtrunks.

Your service provider will issue you with one or a number of DID numbers with which people can

call you on.

Click “Number DIDs” tab and click “New Number DID” button to add a number DID rule:

In this example, if the caller calls your DID number 51097214 the call will go directly to extension

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410, general inbound control will not work with this DID number. If you experience problems

setting inbound DID then please check with your service provider to confirm the exact DID

number that the service provider is passing to theSyn_PBX.

DOD Settings

DOD is also known as direct outward dialing, by specifying the number of an external caller in

theSyn_PBX system, when this caller calls in, this call can be directed to a destination directly

without restriction of time rule or IVR.

Click DOD Settings tab and click New DOD to add a record.

For this example, if the caller 02885337096 calls the office number, the call will go directly to

extension 405.

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4. SYN_PBX Advanced

4.1 Global SYN_PBX Advanced Settings

4.1.1 General

Navigate to web menu Advanced->Options->General.

Here on this page you can configure some global options for all user extensions. In the “Local

Extension Settings” section you can view the below options that can be configured.

Local Extension Settings

 Operator Extension: Choose an extension to be operator extension. When an incoming call

has been directed to voicemail, then by pressing ‘0’ the caller will be put through to the

operator extension.

 Global Ring Time Set(sec): If not specifically configured, an incoming call will ring the

extension for the time given here.

 Enable Transfer: If enabled, the extension users will be able to perform call transfers.

 Enable Attended Transfer Caller ID: Normally if you use feature code *2(This will be

introduced in chapter 4.18) to transfer a call to another extension, the extension user only

sees your extension number as caller ID but not the actual caller ID, by enabling this option

the real caller ID will be passed to the user extension.

 Enable Music On Ringback: If enabled, callers will hear music instead of ringback tone when

calling other extensions.

 Auto-Answer: Auto-answer enables the SYN_PBX to automatically answer the inbound calls

from analog ports.

 Fax Detect Time: If auto-answer enabled, you are able to configure the fax auto detection

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time here.

 Web Dial Auto-Answer: Enable/disable auto answer of the extension numbers while dialing

from Web GUI.

 Record Format: Choose GSM or WAV as the call recording format.

 Call Forward CID: Allow passing the real caller ID to the forwarded number.

 P-Preferred-Identity: The P-Preferred-Identity header is used among trusted SIP entities

(typically intermediaries) to carry the identity of the user sending a SIP message as it was

verified by authentication.

Default Settings for New User

In this section, options are defined for the creation of new extensions. If you have one of the

options enabled, then so will any newly created extensions.

Extension Preferences

The user extension number and system extension number ranges are defined here to avoid any

conflicts within the Syn_PBX system. You can modify these number ranges according to your

requirements.

4.1.2 Global Analog Settings

Global Analog Settings are used for configuring the Syn_PBX system to seamlessly work with the

telephone lines from your telecommunications provider.

Navigate to web menu Advanced->Options->Global Analog Settings.

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Caller ID Detect

These options are used to teach the Syn_PBX system how to detect caller identity(caller ID) from

the PSTN lines on FXO ports.

 Caller ID Detection: Enable/Disable Caller ID Detection.

 Caller ID Signaling: The signaling type applied on the PSTN lines to pass caller ID.

Bell-US—Also known as BellcoreFSK. Used in the Canada, China, Hong Kong and US.

DTMF—Dual Tone Multi-Frequency. Used in Denmark, Finland and Sweden.

V23—Mostly used in UK.

V23-Japan—Mostly used in Japan.

 Caller ID Start: Defines when the caller ID starts.

Ring—Caller ID starts when a ring is received.

Polarity—Caller ID starts when polarity reversal starts.

Polarity(India)—Can be used in India.

Before Ring—Caller ID starts before a ring received.

 CID Buffer Length: The buffer length can be used to store caller ID info.

 Ring Debounce: Sets the minimum time in milliseconds to debounce extraneous ring events.

 DTMF Hits Begin: Sampling matching value of DTMF caller ID digits, you can choose 1 to 5

digits been matched then to consider it as part of the Caller ID.

 DTMF Misses End: Sample matching value of DTMF caller ID digits, you can choose 1 to 5

digits been mismatched then to consider it’s not part of the caller ID.

 Detect Caller ID After: Sets the SYN_PBX system to detect Caller ID after specific rings

received.

General

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 Opermode: Set the Opermode for FXO Ports.

 ToneZone: Select the tone zone of your country.

 Ring Timeout(s): FXO (FXS signaled) devices must have a timeout to determine if it should

hang up before the line is answered. This value can be tweaked to shorten how long it takes

before DAHDI considers a non-ringing line to have hung-up.

 Relax DTMF: Helps DTMF signal detection.

 Send Caller ID After: Certain countries (UK) have ring tones with different ring tones

(ring-ring),which means the caller ID needs to be set later on, and not just after the first ring,

as per the default (1).

 Echo Cancel: Enable/Disable software Echo Cancel algorithm.

 Echo Training: Enabling echo training will cause the PBX system to mute the channel, send

an impulse, and use the impulse response to pre-train the echo canceller so it can start out

with a much closer idea of the actual echo. Value may be "yes", "no", or a number of

milliseconds to delay before training (default = 400). This option does not apply to hardware

echo cancellers.

4.1.3 Global SIP Settings

Global SIP settings allow you to configure some general and advanced options for the IP-PBX

system global SIP preferences. Navigate to web menu Advanced->Options->SIP Settings.

General

 UDP Port: SIP over UDP service port. By default Synway SYN_PBX system uses UDP as SIP

transmission protocol. Port number can be changed here if required.

 TCP Port: By ticking the “Enable” checkbox you can enable global SIP TCP support. To

extension configure page, please refer to chapter 2.7.1.

 TLS Port: By ticking the “Enable” checkbox you can enable global SIP TLS support. To register

a SIP extension over TLS protocol, you’ll have to select TLS transport on the extension

configuration page, please refer to chapter 2.7.1.

 Start RTP Port/End RTP Port: The UDP ports used by Syn_PBX system to carry RTP voice

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stream. Do not change the port numbers or you may encounter audio issue with phone

calls.

 DTMF Mode: The DTMF mode specifies how touch tones will be transmitted to the other

side of the call. Possible values for this field are rfc2833, inband, info, and auto.

 Allow Guest: This setting determines if anonymous callers are permitted to place calls to

theSyn_PBX system. For security precautions please do not enable this option.

 Max Registration/Subscription Time(sec): Maximum allowed time of incoming registrations

and subscriptions (seconds).

 Min Registration/Subscrption Time(sec): Minimum length of registrations/subscriptions.

 Default Incoming/Outgoing Registration Time(sec): Default length of incoming/outgoing

registration.

NAT Support

When the Syn_PBX system is behind a NAT device and needs to communicate to the outside. It

needs to know whether it is talking to someone "inside" or "outside" of the NATted network. For

example, if you are going to deploy remote extensions you have to tell the Syn_PBX system which

network address/addresses are from inside and which are from outside. Below is an example

configuration.

 External IP: Your static public IP address or domain name.

 External Host: This is similar to “External IP” except that the hostname is looked up every

"External Refresh" seconds(default 10’s).

 External Refresh(sec): The refresh interval of the “External Host”.

 Local Network Address: Your local network address/addresses.

Notice:

If you have one-way audio or no audio issue on the remote extensions then this most probably means that NAT

support is not properly configured. Please check your configurations here.

Type of Service

Asterisk supports different QoS settings at the application level for various protocols on both

signaling and media. The Type of Service (TOS) byte can be set on outgoing IP packets for various

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protocols. The TOS byte is used by the network to provide some level of Quality of Service (QoS)

even if the network is congested with other traffic.

 TOS for Signaling Packets: Sets TOS for SIP packets.

 TOS for RTP audio packets: Sets TOS for RTP audio packets.

 TOS for RTP video packets: Sets TOS for RTP video packets.

 COS Priority for Signaling packets: Sets 802.1p priority for SIP packets.

 COS Priority for RTP audio packets: Sets 802.1p priority for RTP audio packets.

 COS Priority for RTP video packets: Sets 802.1p priority for RTP video packets.

 DNS SRV Look Up: Enable DNS SRV lookups on outbound calls.

 Relax DTMF: Relax DTMF handling.

 RTP TimeOut(sec):Terminate call if there is 60 seconds of no RTP or RTCP activity on the

audio channel when we're not on hold. This feature enables the ability to hangup a call in

the case of a phone disappearing from the network, for instance if the phone loses power.

 RTP Hold TimeOut(sec): Terminate call if 300 seconds of no RTP or RTCP activity on the audio

channel when on hold.

 Add ‘user=phone’ to URI: Enable this option if the SIP provider requires ";user=phone" on

URI.

 UserAgent: Allows you to change the user agent string. The default user agent string also

contains the Asterisk version. If you don't want to expose this, change the user agent string

here.

 Premature Media: If enabled, SIP channel will not send 183 SessionProgress for early media.

Before enabling premature media make sure thatprogressinband is configured as never.

 Progress Inband: Sets the SIP channels to use inband signaling or not.

Outbound SIP Registrations

The “Outbound SIP Registrations” configures the register behaviors of Syn_PBX system when

registering as a client to the other SIP servers.

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 Register TimeOut(sec): Retry registration every 20 seconds (default).

 Register Attempts: Number of registration attempts before the SYN_PBX system give up.

Default is 10 and 0 means continue forever.

4.1.4 Global IAX Settings

Navigate to web menu Advanced->Options->IAX2 Settings.

 UDP Port: IAX2 signaling and media port, the default is 4569.

 Bandwidth: Specify bandwidth of low, medium, or high to control which codecs are used in

general.

 Max Registration/Subscription Time(sec): Maximum amount of time that IAX peers can

request as a registration expiration interval (in seconds).

 Min Registration/Subscription Time(sec): Minimum amount of time that IAX peers can

request as a registration expiration interval (in seconds).

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4.2 Virtual Fax

Syn_PBX system has the ability to auto detect incoming faxes and send the received faxes to a

user’s email box. If you don’t wish to send the fax by email then faxes can be saved to a user’s

extension account.

Notice:

Please enable Virtual Fax services on Virtual Fax page first, and then follow the instructions below to configure.

4.2.1 Receive Fax

Syn_PBX system detects incoming faxes from the trunks. To configureSyn_PBX to auto detect

incoming faxes please navigate to web menu Basic->Trunks.

Click on “Edit” to edit the trunk(either analog or VoIP trunk) that you want to configure fax auto

detection on. Find the “Auto Fax Detection” option and tick the checkbox. You’ll see a dropdown

list from which you can select any extensions to direct the detected faxes to.

If you want the SYN_PBX system to send the received faxes to an email address(Fax to Email) then

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please select an extension number starting with “Virtual Fax”. Then navigate to Basic->Extension

page to specify the email address in “Email(Fax/Voicemail)”section.

If you require that the received fax is stored in the SYN_PBX system only then you should select a

virtual fax extension without specifying the email address.

Finally, if you want the incoming fax to be handled by a fax machine, please select the extension

number assigned to the fax machine.

Notice:

If you are configuring Fax to Email, you also have to configure the SMTP service before it will work. Please refer to

chapter 4.3.4.

4.2.2 Send Fax

To send a fax you must first login to the Syn_PBX web interface with an extension number and

the voicemail password for this extension. Before doing this please ensure this extension has the

“Web Manager” option enabled on the extension configure page.

After login, navigate to the Send Fax page.

Enter the fax number and click on “Choose File” to locate the file you are planning to send,

upload the file and then send the fax.

There are some optional options for outbound faxes, please navigate to web menu

Advanced->Virtual Fax.

 Enable: Enable Virtual fax feature for receiving and sending faxes.

 Country Code: Enter your country code here.(Optional)

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 Area Code: Enter your Area Code here.(Optional)

 Outbound CID: Only works if the outbound fax is to be sent through VoIP trunks. The other

side receives your fax with this number.

 Label: Define custom information to be printed to the header of the fax pages.

 Fax Seat: Defines how many users can send fax at the same time.

 DialPlan: A dial plan to send faxes.

4.3 VoiceMail

4.3.1 General Voicemail Options

Voice mail allows callers to leave messages for subscribers (user extensions) of the SYN_PBX

system when they are unable to answer the incoming calls.

VoiceMail Reference

 Max Greeting Time(sec): Maximum voicemail box greeting message duration.

 Dial “0” for Operator: If this option is enabled then callers will be able to dial "0" to transfer

out of voicemail to the Operator.

Voice Message Options

 Message Format: The audio file format to be used for the recording.

 Maximum Messages: The maximum amount of voice messages for each extension.

 Max Message Time(min): The maximum time duration of an individual voicemail message.

 Min Message Time(sec): The minimum time duration of an individual voicemail message.

Default minimum duration is 2 seconds, which means voice messages which are less than

2seconds will be ignored by the SYN_PBX system.

Playback Options

These options are for voicemail message playback.

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 Say Message CallerID: Announce the Caller ID of the caller who left this message before

playing the voice message.

 Say Message Duration: Announce the message duration before playing the voice message.

 Play Envelope: Announce the date, time and caller ID for the voicemail message.

 Allow Users to Review: If enabled, this option will allow users to review the voice message.

4.3.2 Playback Voicemail on the phone

Navigate to web menu Advanced->Feature Codes.

On this page, you’ll find two feature codes that can be used for checking voicemail.

Voicemail Main Menu: *60

Check Extension Voicemail: *61

Dial *60 and you will enter the main menu of voicemail feature, by specifying the extension

number and voicemail password of the required extension then you can check its voicemail and

you can do this for any extension by following the system voice guidance.

By dialing *61 from an extension and entering the voicemail password forthis extension you can

follow the voice guidance to check voicemail of your own extension. Or alternatively, you can

configure some advanced options for your voicemail box.

4.3.4 Voicemail to Email

To send received voicemail messages to the user’s email box, you need to configure SMTP

support, Email format and specify email addresses for the extension users.

Step1:

SMTP Settings

Navigate to web menu: Advanced->SMTP Settings.

Define an email account to be used by the system which will send emails with voicemail

messages attached to the extension users’ email

boxes.

 SMTP Server: SMTP server domain, for example: smtp.gmail.com, smtp.tom.com.

 Port: Default SMTP service port is 25, but if you are using SSL/TLS then please use port 465.

 SSL/TLS: Encrypts a communication channel between the Syn_PBX system and the SMTP

server.

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 Enable SMTP Authentication: If your SMTP server requires authentication then please

enable this option and configure the following.

 Username: The email account.

 Password: The password for this email account.

 Send Test: Click “Send Test” to send a test email to see if SMTP is working correctly. If it is

working then you’ll receive an email sent by the SYN_PBX system.

Step 2:

Email Settings

Navigate to web menu: Advanced->Voicemail->Email Settings.

On this page you can define the email content that will be sent to the extension users’ email

boxes.

 Attach voicemail to email: If enabled, the system will send any voice message files received

to the extension users’ email box.

 Sender Name: Alias for the SMTP email account.

 From: The email account from SMTP settings.

 Subject: The subject of the email sent bySyn_PBX system.

 Message: The content of the email, describes the details of the voicemail message received.

 Template Variables: These variables can be used to acquire details of the voicemail messages,

which can then be used in the message field to compose the email content.

Step3:

Email Address

Go to the extension details for the user and specify the email address where messages for this

user should be sent.

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Once these 3 configuration steps are complete, if user extension 401 receives a new voicemail

message then the SYN_PBX system will send this voicemail message to example@gmail.com.

4.3.5 Playback Voicemail from Web GUI

An extension user can login to the web interface with their extension number and voicemail

password if “Web Manager” option is enabled on their extensions.

Navigate to Voicemail List page.

Here on this page you can see all newly received voice messages displayed.

By clicking “Play” button you will be presented with a dialog box that gives you two options to

playback this message.

By clicking button you can playback this message directly from the web interface. By

selecting an extension number and clicking on the “Play” button you can playback this message

from the selected extension.

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4.4 Conference

Conferences allow two or more callers to be joined together so that all parties on the call can

hear one another. Conferences are also referred as Conference Bridges or Conference Rooms.

OnSyn_PBX system, you can create up to 20 conference rooms. There are 3 default conference

rooms preconfigured for you.

4.4.1 Static Conference

Navigate to web menu Advanced->Conference. You can click “New Conference” button to add a

new conference room or click “Edit” button on the existing conference room to change the

properties.

Conference number

 Room Extension: Call this extension number to enter the conference room.

Conference Password

 Guest Password: If callers use this password to enter the conference then they are ordinary

participants.

 Administrator Password: If callers use this password to enter the conference then they are

administrators and have advanced conference menu features such as inviting people to

participate in the conference.

Conference Options

 Conference DialPlan: Conference admin can use this dialplan to invite other participants.

 Play hold music for first caller: Plays the hold music for the first participant in the conference

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until another participant enters the conference.

 Enable caller menu: Check this option to allow the conference admin to access the

conference menu by pressing “*” on the phone.

 Announce Callers: Announce all the participants in the room when a new participant enters

the conference room.

 Record Conference: Record this conference(Recording format is wav). The recorded

conference can be searched within Report->Record List->Conference page. Please see

chapter 6.3.2.

 Quiet Mode: If this option is checked then the system will not give any announcement when

participants enter or leave the conference.

 Close the conference when last administrator exits: If this option is checked then the

conference will be terminated when the last administrator exits.

 Leader Wait: Wait until the conference leader(administrator) enters the conference before

starting the conference.

4.4.2 Dynamic Conference

Syn_PBX system allows you to press a key sequence(feature code) to create a conference during a

live call.

Please navigate to web menu Advanced->Feature Codes. You can see the feature codes available

for conference feature.

 Invite Participant: When in a static conference room or a dynamic conference room, if the

conference administrator presses 0 they will get a dial tone to invite others to participate in

this conference.

 Create Conference: During a live call the extension user can press *0 to create a dynamic

conference room. The other side will automatically enter the conference as an ordinary

participant while the extension user who created this conference will be requested to enter

the conference password to enter.

 Return to conference with participant: While using the conference menu to invite other

people, you can dial ** to return to the conference with invited party.

 Return to conference without participant: If the invited party doesn’t want to participate in

the conference you can press *# to return to the conference without the invited party.

Notice:

After a dynamic conference is created, in reality you have entered a static conference room (by default 900 is the

first available conference room). You are able to use conference admin menu to invite others to the conference

also others can dial 900 to enter this conference.

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4.5 Music Settings

Music Settings, or Music On Hold(MOH) as it is more commonly known on an SYN_PBX system

allows audio files (such as WAV or MP3 files) to be uploaded to the SYN_PBX system and played

back when a caller is placed on hold or is waiting in a queue.

Navigate to web menu Advanced->Music Settings.

 Music On Hold Reference: Audio files in this selected folder will play to the party which is on

hold.

 Music OnRingback Reference: Audio files in this folder will be played instead of playing

ringback tone to the caller.

 Music On Queue Reference: Audio files in this folder will be played when the caller is waiting

in a call queue.

There are 10 folders for music files, by default the first 3 folders are preloaded with music files

which you may wish to choose. However, if you want to upload your own audio files please click

“Music Management” tab.

In the Music Management section, you can select a music folder and click “Load” button to check

which audio files are inside this folder. By clicking “Delete” button you can delete the existing

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audio files.

In the Upload Music File section, you can select a music folder and browse your PC file system to

select your preferred audio file and click “Upload” button to upload the audio file. If there are

more than one audio file in the same music folder, they will be played at random.

Notice:

Syn_PBX system can adopt MP3, wav(16bit, 8000Hz, mono), gsm, ulaw and alaw audio file format.

4.6 DISA

Direct inward system access(DISA) allows an outside caller to dial directly into the PBX system and

access the system's features and facilities remotely.

This is useful if you want people to be able to for example take advantage of the low rate for

international calls that you have available on your system, or to allow outside callers to be able to

use the paging or intercom features of the system. Always protect this feature with strong

password/passwords, the passwords need to be set on Advanced->Pin Sets page which will be

introduced in Chapter 4.13.

Navigate to web menu Advanced->DISA. Click on “New DISA” button to create a new DISA call

target.

 Name: Alias of the DISA call target.

 PIN Set: A set of pin codes used to authorize all callers using the system features and

facilities.

 Without PIN: If enabled, callers will not be required to enter any pin code to be able to use

the system features (Not recommended).

 Record in CDR: The pin code that is used will be stored into call logs and can therefore be

traced on Report->Call Logs page.

 Response Timeout(sec):The maximum waiting duration before hanging up if the dialed

number is incomplete or invalid. Defaulted 10 seconds

 Digit Timeout(sec):The maximum interval time between digits when typing extension

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number. Defaulted 5 seconds.

 Extension for this DISA(Optional): If you want to access DISA by dialing an extension, you can

define an extension number for this DISA.

 Select DialPlan: Select a dial plan for this DISA so callers will be able to make outbound

phone calls using the trunks on the SYN_PBX system.

Notice:

After a new DISA is created, it can be included in the inbound control section as a call destination. But this is not

recommended as it is not safe because all callers can possibly access DISA functionality. A better option is to

configure DOD settings (Chapter 3.3.6) for the numbers which you want to be able to access DISA.

4.7 Follow Me

The Follow Me feature allows you to set a list of numbers that you may possibly be contacted on.

Therefore, if someone calls your extension and you are not available then follow me will work

through the list calling each of the numbers in turn until you are contacted or the list is

exhausted.

To configure follow me, navigate to web menu Advanced->Follow Me. Click on “New Follow Me”

to configure follow me for an extension.

 Extension: Select the extension number which will be configured with follow me.

 Ring lasting for _20_ seconds: Define how long to ring the extension before the call is

forwarded out. By default, this is 20 seconds.

 Follow Me List: The list of numbers to forward the calls to. Each line is written with the

format “number,time”, “number” is one of the number to forward the calls to, “time”

defines how long to ring this number. They are separated with a comma without space.

Numbers are called in sequence.

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4.8 Call Forward

4.8.1 Configure from the Web

This feature allows calls to an extension to be automatically forwarded to a specific internal

extension or external phone number.

Before configuring call forward you can enable the SYN_PBX system to play a voice prompt before

the call is forwarded. This voice prompts can be recorded or uploaded from the Inbound

Control->IVR Prompts page.

Once the voice prompt file is ready you can navigate to web menu Advanced->Call Forward and

enable the system to play back the voice prompt before the incoming call is forwarded.

After the voice prompt is set, click “New Forward” button to set call forward for an extension.

 Always: Unconditionally forward the incoming calls.

 Busy: Forward the incoming calls only if the extension is busy.

 No Answer: Forward the incoming call only if the extension didn’t answer.

 Ring lasting for ____ seconds: Only configurable for “No Answer” option. It defines how long

to ring the extension before forwarding if the extension didn’t answer.

Notice:

1. If you are forwarding a call to an external phone number then please ensure that you add a prefix in front of

the number if your system requires a prefix to dial out.

2. The forward condition “Always” is mutually exclusive to “Busy” and “No Answer”.

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4.8.2 Configure from the Phone

Navigate to web menu Advanced->Feature Codes.

You’ll see feature codes for call forward as follows:

With these feature codes, you can activate or deactivate call forward directly from your phones

without configuration on the Web GUI.

For example, a Syn_PBX requires prefix 9 to call outbound, and the number you want to forward

the calls to is 85337096.

 Activate always call forward: Dial *71985337096, press 1 to confirm.

 Deactivate always call forward: Dial *071.

 Activate call forward on busy: Dial *72985337096, press 1 to confirm.

 Deactivate call forward on busy: Dial *072.

 Activate call forward no answer: Dial *73985337096, press 1 to confirm.

 Deactivate call forward no answer: Dial *073.

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4.9 Call Transfer

Call Transfer is used to transfer a call in progress to some other destination. There are two types

of call transfer.

 Attended call transfer - Where the call is placed on hold, a call is placed to another party,

and a conversation can take place privately before the caller on hold is connected to the new

destination. It is also referred to as "Supervised Call Transfer".

 Blind call transfer - Where the call is transferred to the other destinations without

intervention (the other destination could ring out and may not be answered for instance).

Navigate to web menu Advanced->Feature Codes. You’ll see the feature code for call transfer as

below:

 Blind Transfer: In a live call, an extension user can press # key and the SYN_PBX system

prompts “Transfer”, you then enter the number to transfer to, this call will be transferred

instantly and the user can hangup. If the transferred number doesn’t answer this call then it

will ring back to the extension user.

 Attended Transfer: In a live call, extension user can press *2 and the SYN_PBX system

prompts “Transfer”, you then enter the number to transfer to, after someone answers your

call, you can introduce this call and hangup at which point the call is transferred.

 Disconnect Call: In an attended transfer if the other side doesn’t want to take the call to be

transferred, you can press * to disconnect with them and get back to the caller.

 Timeout for answer on attended transfer(sec): In an attended transfer, if the third party rang

for 15 seconds without answering, the extension user will go back to the caller and the

transfer is terminated.

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4.10 One Number Stations

One Number Stations is an innovative SYN_PBX feature unique to Synway SYN_PBX. With one

number stations feature, you can have the same extension number in several different locations.

One number stations feature can put several extension numbers in the same “group”, a main

number can be selected from the members and when an incoming call is made to the main

number, it will ring all the member extensions including the main number. Any extension with the

group calling other extensions will display only the main number.

Navigate to web menu Advanced->One Number Stations. Click “New One Number Stations”

button to create a one number stations group.

Select the extensions from the “Extensions” column to the “ONS Group Members” column. In the

“Main Extension” dropdown list select an extension to be the main extension number. Next click

“Save” and you’ll have a new one number stations group.

In this example, no matter whether 407, 408 or 409 makes a call, other extensions only see the

calling number as extension 407, while any calls made to 407 will result in all 3 extensions

ringing.

As you can see on this page there’s a feature code Switch Station available.

This feature code is used to switch extension during a phone call. For example, if an inbound call

called extension 407 and the one number stations member 408 answered this call, you can press

*1 from extension 407 or 409 to switch this live call to 407 or 409, then 408 will be disconnected.

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4.11 Paging and Intercom

The Paging and Intercom feature allows you to use your phone system as an intercom system,

provided that your endpoints (phone devices) support this functionality. The Paging and Intercom

feature allows you to define a number (just like an extension or Ring Group number) that will

simultaneously page a group of devices. For example, in a small office, you might define a paging

group that allows any user to dial 699, allowing them to page the entire office. You can also use

the feature code *50/*51 to page/intercom a single extension, by dialing *50/*51 followed by

the extension number.

Navigate to web menu Advanced->Paging and Intercom. Click “New Paging and Intercom” button

to add a new paging group.

 Paging Extension: The extension number for this paging group, by calling this extension

number you can reach the group members.

 Description: Description of this paging group.

 Duplex: If enabled, the group members can talk back to the caller.

By calling the paging extension number, all group member phones will auto answer in speaker

mode (requires that the IP phones support auto answer feature), the caller can now make a brief

announcement to the group members.

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4.12 Web Extensions

Web Extension is a new feature that makes use of WebRTC technology. You can use any web

browser that supports WebRTC to register an extension number to yourSyn_PBX system without

any plugins.

To register the first Web extensions please follow the steps below:

Step 1:

Create a Web Extension

To create a web extension, navigate to web menu Advanced->Web Extensions. Click on “New

User” button to add a new web extension.

 Name: Username of this web extension.

 Extension: Extension number of this web extension.

 Password: Password for registration of this web extension.

 Outbound CID:Only works if the call was placed out through VoIP trunks.

 DialPlan: Defines which type of numbers the web extension can dial.

 Transport: WS or WSS.

 WS: WS (WebSocket) Protocol which is an independent TCP-based protocol providing

full-duplex communication channels over a single TCP connection. The WebSocket protocol

was standardized by the IETF as RFC 6455 in 2011, and the WebSocket API in Web IDL is

being standardized by the W3C.

 WSS: WSS (WebSockets over SSL/TLS), like HTTPS, WSS is encrypted and we strongly

recommend the secure wss:// protocol over the insecure ws:// transport. A variety of

attacks against WebSockets are almost impossible if the transport is secured.

Step 2:

Upgrade Web extension patch

As you can see, web extensions use different protocols for signaling and media (WS/WSS) and

they are not ordinary SIP/IAX2 extension that can use IP phones or softphones to register so must

be treated differently.

Download WebRTC patch from the links below:

U20V2: http://Synway.com/html/U20_V2.html

U50V2: http://Synway.com/html/U50_V2.html

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Step 3:

After completing the upgrade process (see chapter 8.6) you can access the WebRTC extension

(192.168.1.254 should be your SYN_PBX IP address) you will see the web extension register

interface. Please complete the register credentials as below:

Next, press Enter and the web extension will be registered and is ready for phone calls just like

any other standard extension.

WebRTC can even be adapted to the enterprise website which can help an enterprise serve their

customers with direct voice communication via their website. For more advanced WebRTC

settings please refer to the WebRTC manual.

4.13 Pin Sets

Pin sets can be used to secure your SYN_PBX system phone services and in particular for

outbound dial rules and DISA.

Navigate to web menu Advanced->PIN Sets. Click on “New PIN Set” button to create a collection

of PIN codes.

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Each line is a PIN code, press Enter to add the next PIN code without any symbols.

4.14 Call Recording

Syn_PBX system has built-in ability to record calls. No additional software is required for

recording calls. When Syn_PBX system records a call, both sides of the call are recorded and

written out to a file for playback on a computer. Call recording can be used to ensure call quality,

or to keep calls for later review. Syn_PBX provides the ability to record all of the calls, or to

selectively record calls.

4.14.1 Record All Calls

Navigate to web menu Advanced->Call Recording. Click “New Call Recording” to activate call

recording for the extensions you want calls to be recorded.

 Extension: Select the extensions which you want their calls to be recorded.

 Always Recording: If enabled, all calls from the above selected extensions will be recorded

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regardless when the calls were made and received.

 Start Time, End Time, Start Day, End Day: If Always Recording is unnecessary then you can

specify which time durations in a week to record all calls from the above selected

extensions.

 Inbound Record: Enable to record all inbound calls.

 Outbound Record: Enable to record all outbound calls.

The recordings can be searched on Report->Record List->Call Recording page. Please see chapter

6.3.1.

4.14.2 One Touch Recording

One Touch Recording is also known as Record on Demand. It allows users to record phone calls

selectively.

Navigate to web menu Advanced->Feature Code. Here on this page you can see the one touch

recording feature code as below:

In a live call conversation, an extension user can use feature code *1 to record this call. With this

feature, you don’t have to configure recording all calls for the extensions which may cause heavy

system resource use if some call recordings are not required.

The one touch recordings can be searched from Report->Record List->One Touch Recording page.

Please see chapter 6.3.3.

4.15 Smart DID

Syn_PBX system has the ability to route an inbound call directly to an extension if the extension

had previously tried to call the number but the call was unanswered. It is convenient for the

called party to make a call back and be directly routed to the extension that called them without

going through the IVR menu or reception desk.

Navigate to web menu Advanced->Smart DID. Tick the “Enable” checkbox to enable Smart DID

functionality.

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There is a default Smart DID rule which enables all outbound calls to be monitored by the Smart

DID feature. If the call is not answered by the called party, then the called number will be stored

into the Asterisk database with the extension number which made this call.If the called party

does make a callback to the SYN_PBX system then the call can automatically be directed to the

extension number.

If you don’t want all outbound calls monitored by Smart DID, you can modify the existing rule or

click “New Smart DID Rule” to add your custom rule/rules. An example of this is detailed below:

 Pattern: Defines the number format which would be dialed.

 Strip: Remove some digits from the front of the dialed number.

 Prepend: Prepend some digits in front of the dialed number after manipulated by the “Strip”

option.

The numbers to be dialed will start with prefix 17951 and if they call back, the expected numbers

will have +86 in front of them instead of the 5-digit prefix 17951. In such a situation, the

outbound and inbound numbers are not the same, you’ll need the “Strip” and “Prepend” options

to manipulate the dialed numbers to make sure it can match the “same” number when it calls

back. If the numbers to be called and the numbers to be received are the same, then you don’t

have to configure these 2 options. Alternatively, you can configure only one of these 2 options, it

will all depend on your actual requirements.

For example, the extension user 401 wishes to call 85337096, and the carrier requires a prefix

17951 to ensure the rate is much cheaper. The user will dial 1795185337096 to place this call. If

the called party misses this call then the SYN_PBX system will store this number +8685337096

with extension number 401 into its database. Later on, if the called party tries to call back, the

SYN_PBX system gets +8685337096 as the caller ID and matches this from its database, once

successfully matched, this call will be automatically directed to extension 401.

Notice:

1. The records for Smart DID functionality in the system database will be erased every day at midnight. This means

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this is a dynamic effective feature and is only designed to handle callbacks made within the same day as the

original call.

2. In the “Pattern” field, patterns can be used in the same way as the patterns used to manipulate dialed number

in the dial rules. Please refer to chapter 3.2.1.

3. Due to the mechanism of how Asterisk works, at this time Smart DID only works with VoIP trunks and does not

work with FXO or GSM/ WCDMA trunks.

4.16CallBack

Callback is to allow a company employee who needs to make a call from their personal phone to

call the SYN_PBX, the SYN_PBX calls them back and the cost of any future outbound calls are at

the companies expense.

Navigate to web menu Advanced->Call Back.

 Enable: Check the checkbox to enable call back feature.

 Strip: The received caller ID might have some additional digits in front of it and it will not be

possible for you to call back directly, you can specify here to remove some digits before

calling back.

 Prepend: After the number has been manipulated by the “Strip” option, you can use this

option to add some extra digits in front before calling back.

 DialPlan: Choose an appropriate dial plan to make sure the SYN_PBX system has the

permissions for outbound calling.

Click “New Callback Number” to add a call back number.

 Callback Number: The number which will be used to call into the SYN_PBX system and will

be handled by the Callback feature.

 Destination: An extension or another call destination which will be used to call the callback

number.

In the above example, if the caller 13880424687 called the SYN_PBX system, SYN_PBX will

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disconnect this call and make a call back to this number using extension 410.

In the call back destination field you can even set the destination to a conference, call queue or

DISA, so the callers can access these functionalities all at the companies expense.

4.17 Phone Book

The phone book on theSyn_PBX system is similar to a contact list on acellular phone. You can add

the contacts to the SYN_PBX system from Advanced->Phone Book page.To do this Click “New

Contact” to create a new contact record.

 Name: Contact name.

 Phone Number: Phone number of the contact.

 Speed Dial: Speed dial number which can be used to call this contact from another

extension.

After contacts have been created they will be listed here on this page.

Here on this page you also have some additional advanced options for the phone book and LDAP

configurations.

 Import: You can import a contact list from .txt or .csv files.

 Export: Export the current contact list as .csv file.

 Delete All: Delete all contacts.

 Sync LDAP: Synchronize the contacts to an LDAP server.

 The prefix for speed dial: Using this feature code with the speed dial code of a contact you

can call the contact without knowing their exact number.

 Filter: Search contacts by contact name, phone number or speed dial code.

 Create Contact: Create a new contact record.

 Delete Selected: Delete the selected contacts.

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 Call: Assign an extension to call this contact.

 Edit: Edit the information of this contact.

 Delete: Delete this contact.

4.18 LDAP Server

4.18.1 LDAP Server Settings

LDAP(Lightweight Directory Access Protocol) is an open, vendor-neutral, industry standard

application protocol for accessing and maintaining distributed directory information services over

an IP network. An LDAP server has been embedded into Syn_PBX IP PBX which is mainly used to

centralized and manage the phonebook. LDAP server has generated the phonebook based on

created extensions by default.

Navigate to web menu Advanced->LDAP Server.

 Enable: Enable/Disable LDAP Service.

 Username: Define the username of the server administrator (e.g.: manager). This setting will

be used on the IP Phone.

 Password: Define the password of the server administrator. This setting will be used on the

IP Phone.

 Domain: Define a domain for the LDAP server (e.g.: ldapdomain.com).This setting will be

used on the IP Phone.

 Organization: Define an organization to describe the members recorded by LDAP (e.g.:

Synway.ltd). This setting will be used on the IP Phone.

 Port: LDAP service port, the default port number is 389.

4.18.2 Synchronize Contacts with LDAP Server

Navigate to web menu Advanced->Phone Book. Click on the “Sync LDAP” button to synchronize

contacts with LDAP server.

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4.18.3 LDAP Client Settings

After enabling the LDAP server, you need configure a client. For example: SYNWAY IP Phone.

Open the web interface of the IP Phone on your browser, navigate to web menu Phone->Remote

Contact->LDAP Settings.

 LDAP: Select LDAP server to connect

 Display Title: Define a title for the LDAP server, this will be displayed on the phone’s screen.

 Version: Select the LDAP Version. Default is Version 3.

 Server Address: Configure the server domain or IP

 Server Port: Configure the server port. Default is 389.

 Authentication: Select authentication method, including None, Simple, Digest-MD5 and

CRAM-MD5.

 Line: Select SIP lines for outbound calls with different LDAP servers

 Username/Password: Configure the username and password for the LDAP server. (Please

refer to the settings of LDAP server, username format: cn=Syn_PBXV2, dc=ldapdSynway.com.

 Search Base: Configure the position where the search begins. (This corresponds to the

domain setting of LDAP server.)

 Enable Calling Search: You can search or update the other party’s name in the call if you

enable calling search.

 Telephone/Mobile/Other/Display Name: Configure the contact information, including

display name, telephone number, mobile number and other number.

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4.19 Feature Codes

Feature codes allow you to set the special codes that users can dial to access various features.

Navigate to web menu Advanced->Feature Codes.

Call Parking

A Parking Lot allows anyone who has received a call to park the call on an extension, allowing any

other user to access the parked call. Typically, you receive the call, transfer it to extension 700,

and then listen as the system tells you where you can pick up the call (usually extension 701).

Anyone else on yourSyn_PBX system can now dial 701 to pick-up the parked call.

A call can be parked for a maximum of 45 seconds as per the definition of “Call Parking Time”, if

nobody picks this call up then it will go back to the extension which parked it.

The “Enable Call Park BLF Notification” enables the parked extensions 701-720 to be monitored

by BLF keys, so if there’s a call that is parked, the extension user will be able to see it from the BLF

panel.

Pickup Call

Pickup call option allows users to pick up calls that are not directed to them by dialing a feature

code *8 or **.

“Pickup Extension: *8 ” has already been introduced in chapter 2.7.1, as it’s related to the

pickup group option of the extension settings.

While “Pickup Specified Extension: ** ” can help pickup a call on any ringing extension. Dial

** followed by the extension number and you can pickup a call on a ringing extension if it is in

the same pickup group as your extension or not.

Transfer

Please see chapter 4.9.

One Touch Recording

Please see chapter 6.3.3.

Call Forward

Please see chapter 4.8.2.

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Do Not Disturb

With the Do Not Disturb(DND) feature enabled, an extension can make outbound phone calls but

inbound calls to the extension cannot be made.

If an extension user of theSyn_PBX system dials *74 from their phone, the system will play a beep

sound to indicate DND has been activated.

To disable DND, simply dial *074, another beep sound will play and DND has been deactivated.

Spy

Call Spy allows users to dial the spy feature codes followed by an extension number to listen to

the call conversation in real-time.

 Normal Spy: For example, extension 410 is talking to someone on the phone, you can dial

*90410 to listen to their conversation, however, neither speaker will be able to hear you.

 Whisper Spy: Whisper spy is also known as coaching. For example, a new employee is talking

to the customer on the phone, their supervisor can dial *91 followed by the employee’s

extension number to listen to their conversation. The supervisor can talk to the new

employee only without the customer hearing the conversation.

 Barge Spy: Barge spy is similar to an instant 3-way conference call. While an extension user is

talking to someone else on the phone, you can dial *92 followed by their extension number

to talk to both of the speakers.

Notice:

Before call spy can be used, you have to make sure the extensions to be spied, have the “Allow Being Spied” option

enabled on extension settings page.

Black List

Black list feature allows you to create a list of numbers that are not allowed to call in to

theSyn_PBX system.

Any extension user can dial *75 and follow the voice prompts to add the numbers to theSyn_PBX

system black list.

To remove numbers from black list, you can dial *075.

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Voicemail

Please see chapter 4.3.2.

Conference

Please see chapter 4.4.2.

Call Queues

Call queue agents can dial *95 to suspend their extension temporarily, new calls will not be

distributed to their extensions, until they dial *095 to resume.

Others

 Intercom: The intercom feature code allows you to intercom one extension only. You don’t

have to create a “Paging and Intercom” group for only one extension if you intend to

intercom with only that extension.

 Paging: The paging feature code allows you to page one extension only. It’s the same as the

intercom feature code, the only difference between paging feature code and intercom

feature code is by using intercom feature code both sides can talk to each other but using

paging feature code, only the caller can talk to the called party.

 Directory: Directory is also known as dial by name. Extension users can dial *3 and follow

the voice prompts to enter the first 3 letters of another extension user’s first or last name

and then make a call to an extension number without knowing its extension number.

 External Paging: A loudspeaker can be connected to the 3.5mm Audio Out interface on the

back panel of the SYN_PBX U100, by dialing *911 you are able to do a paging call to the

loudspeaker to make an instant announcement.

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5. Network Settings

5.1 Network Basic

5.1.1 IPv4 Settings

Syn_PBX system supports static IP, DHCP and PPPoE for WAN connection, while on LAN port only

static IP is supported. If you are configuring your WAN connection as static IP or DHCP, ensure

WAN and LAN IP addresses are not in the same network.

Static

Navigate to web menu Network Settings->Network->IPv4 Setting.

By default, Syn_PBX has been preconfigured with a static IP address of 192.168.1.100 and

192.168.10.100 on WAN and LAN interfaces respectively. If you want to use a static IP then

configure required address here and include the address, netmask, gateway and DNS given by

your ISP or network administrator.

For the LAN interface, you can specify 2 additional virtual IP addresses. These can be used to

access other networks from the LAN port.

DHCP

If your Internet connection automatically provides you with a usable IP address, you can select

“DHCP” on the WAN interface.

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If DHCP is selected then the WAN interface will not be configurable as it obtains all its network

parameters from the DHCP server. DHCP should be used cautiously as all IP extensions register to

the SYN_PBX system through the WAN interface and as DHCP addresses can change and IP

extensions need to know the address of the SYN_PBX at all times. It is best practice to configure

WAN address with a Static IP.

PPPoE

Syn_PBX can be connected to the network via ADSL modem by means of Point-to-Point Protocol

over Ethernet (PPPoE)dial-up. In such a situation, extensions will subscribe to the SYN_PBX

system through the LAN port, while WAN port can be used for remote extensions.

If PPPoE is set, you have to specify the username and password given by your ISP and the

SYN_PBX system will dial-up to the ISP and once successfully connected, you will have Internet

access on the WAN interface.

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LAN port connects to your local network for internal IP extensions to register. If necessary, you

can change LAN IP to suit your local network.

5.1.2 IPv6 Settings

IPv6(Internet Protocol Version 6) has been in development for nearly two decades. Now the

next-generation protocol is ready to replace IPv4 and assume its place as the back of the

Internet.

Today, major Internet service providers (ISPs), home networking equipment manufacturers, and

web companies around the world are permanently enabling IPv6 for their products and services.

Many organizations, institutions and universities have deployed their own networks on IPv6.

To be able to deliver VoIP calls over IPv6(SIP over IPv6), you can configure Syn_PBX system with

IPv6 addresses to be able to deploy it in your IPv6 network infrastructure.

To do this, navigate to web menu Network Settings->Network->IPv6 Settings.

Specify your IPv6 network profile here and you will be able to connect Syn_PBX to your IPv6

network infrastructure.

5.1.3 VLAN Settings

With a layer-3 switch you can configure VLAN on Syn_PBX system to divide the VoIP and data

traffic. Voice VLAN can ensure that phones remain working even when the data network is

congested.

To set VLAN, navigate to web menu Network Settings->Network->VLAN. As you can see here on

this page, you are able to configure 4 VLANs, 2 each for WAN or LAN port.

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Ensure VLAN IPs for VLAN1 and VLAN2 of WAN and LAN interfaces are in several different

network segments.

5.2 Static Routing

Static Routing is a form of routing that occurs when a router uses a manually-configured routing

entry, rather than information from a dynamic routing protocol to forward traffic.

Navigate to web menu Network Settings->Static Routing. Click “New Static Routing” to add a new

routing record to the system.

 Destination: Set the IP address of destination host or network address. E.g.222.209.4.1,

192.168.10.0.

 Gateway: Set the gateway address.

After the new record has been manually created you can see it listed here on this page.

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You can click “Edit” button to edit one of the items, or you can delete the item by clicking the

“Delete” button.

Click the “Routing Table” tab and you’ll see a detailed list of all the system routing rules, including

default and custom ones.

5.3 VPN

VPN(Virtual Private Network) is mainly used for setting up long-distance and/or secured network

connections. When used onSyn_PBX, all phone calls made and received are encrypted so it

secures your remote offices/extensions' phone services. Built-in VPN Server onSyn_PBX series is

an easy way to set up a secured connection between other SYN_PBXs or IP phones. You don't

need to build a dedicated VPN server or buy a VPN router. This is also a workaround to avoid

firewall issues when configuring remote VoIP client such as SIP protocol which is notoriously

difficult to pass through a firewall due to its random port numbers to establish connection.

Syn_PBX IP PBX supports four varieties of VPN, they are L2TP, PPTP, OpenVPN and IPSec.

5.3.1 L2TP VPN

L2TP VPN Server

Layer 2 Tunneling Protocol (L2TP) is a tunneling protocol. It does not provide any encryption or

confidentiality by itself. Rather, it relies on an encryption protocol that it passes within the tunnel

to provide privacy. Here on theSyn_PBX system we use IPSec to do the encryption.

To configure your L2TP server, navigate to web menu Network Settings->VPN Server. Check the

radio button of L2TP to configure L2TP VPN server.

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 Enable: Tick the checkbox to enable L2TP VPN server.

 Remote Start IP, Remote End IP: L2TP VPN remote network IP range, between start IP and

end IP there must be less than 10 available IP addresses.

 Local IP: L2TP VPN local server IP address.

 Primary DNS: Primary DNS for VPN connection.

 Alternate DNS: Alternative DNS for VPN connection.

 Authentication Method : Select the authentication method: chap or pap.

pap: Password Authenticate Protocol, PAP works like a standard login procedure; it uses

static user name and password to authenticate the remote system.

chap: Challenge Handshake Authentication Protocol

CHAP takes a more sophisticated and secure approach to authentication by creating a

unique challenge phrase (a randomly generated string) for each authentication.

 Debug: Tick to enable debug for L2TP VPN connection, debug info will be written into system

logs.

 IPSec: Enable IPSec encryption for L2TP VPN server.

 IPSec Local IP:Syn_PBX WAN IP which can access Internet.

 IPSec Password: Define a password for IPSec VPN client to authenticate.

Notice:

If theSyn_PBX system is behind NAT, you need to open ports 500, 4500 and 1701 on the router/firewall.

For the VPN client to connect you’ll need to create a VPN user account.

Click “VPN User Management” tab and click “New VPN User” button to add a VPN user account.

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Now the L2TP VPN client can connect to the L2TP VPN server.

L2TP VPN Client

For example, in the branch office you are going to connect another SYN_PBX system to the head

office usingL2TP VPN.

Navigate to the web menu Network Settings->VPN Client. Check the radio button of L2TP to

configure L2TP VPN client.

 Enable: Tick to enable L2TP VPN client.

 Server Address: L2TP server public IP.

 Username: L2TP VPN user name given by the VPN server.

 Password: L2TP VPN user password given by the VPN server.

 IPSec: Enable IPSec support.

 IPSec Local IP:Syn_PBX WAN IP Address that can access the Internet.

 IPSec Password: Set according to the password specified on the server.

 Default Gateway: All traffic goes through the L2TP VPN connection.

Notice:

If connection is successfully established, the system will display as follows:

Status: L2TP client VPN remote IP address 172.16.0.1

L2TP client VPN local IP address 172.16.0.x (An IP address between 172.16.0.2 and 172.16.0.9)

5.3.2 PPTP VPN

The Point-to-Point Tunneling Protocol (PPTP) uses a control channel over TCP and a GRE tunnel

operating to encapsulate PPP packets. The intended use of this protocol is to provide security

levels and remote access levels comparable with typical VPN products.

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PPTP VPN Server

To configure your PPTP Server, navigate to web menu Network Settings->VPN Server. Check the

radio button of PPTP to configure PPTP VPN server.

 Enable: Tick the checkbox to enable PPTP VPN server.

 Remote IP: PPTP VPN remote network IP range, there must be 10 or less available IP

addresses between start IP and end IP.

 Local IP: PPTP VPN local server IP address.

 Primary DNS: Primary DNS for VPN connection.

 Alternative DNS: Secondary DNS for VPN connection.

 Timeout(sec): Session timeout for PPTP tunnels.

 Authentication Method: Choose method/methods for the authentication of the VPN clients.

chap: Challenge Handshake Authentication Protocol

CHAP takes a more sophisticated and secure approach to authentication by creating a

unique challenge phrase (a randomly generated string) for each authentication.

pap: Password Authenticate Protocol PAP works like a standard login procedure; it uses

static user name and password to authenticate the remote system.

mschap: MS-CHAP is the Microsoft version of the Challenge-Handshake Authentication

Protocol.

mschap-v2: Microsoft Challenge Handshake Authentication Protocol version 2 (MS-CHAP v2),

this provides stronger security for remote access connections.

 Enable mppe128:Microsoft Point-to-Point Encryption (MPPE) encrypts data in Point-to-Point

Protocol (PPP)-based dial-up connections or Point-to-Point Tunneling Protocol (PPTP) virtual

private network (VPN) connections with 128-bit key.

 Debug: Tick to enable debug for PPTP VPN connection, debug information will be written

into system logs.

You will need to create a VPN user account for a VPN client to be able to connect to the VPN

Server.

To create an account, click “VPN User Management” tab and click “New VPN User” button to add

a VPN user account.

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Now the PPTP VPN client will be able to connect to the PPTP VPN server.

Notice:

If the Syn_PBX system is behind NAT, you will need to open ports 1723 on the router/firewall.

PPTP VPN Client

To create your VPN client at the branch office site, open the Syn_PBX web GUI and navigate to

web menu Network Settings->VPN Client. Check the radio button of PPTP to configure PPTP VPN

client.

 Enable: Tick to enable PPTP VPN client.

 Enable 40/148-bit encryption for MPPE: Tick to enable 40-bit key (standard) or 128-bit key

(strong) MPPE encryption schemes.

 Server Address: PPTP VPN server public IP.

 Username: PPTP VPN user name given by the VPN server.

 Password: PPTP VPN user password given by the VPN server.

 Default Gateway: All traffic goes through the L2TP VPN connection.

Notice:

If connection is successfully established the system will display:

Status: Local IP address 172.16.0.x (An IP address between 172.16.0.2 and 172.16.0.9)

Remote IP address 172.16.0.1

5.3.3 OpenVPN

OpenVPN is an open-source software application that implements virtual private network (VPN)

techniques for creating secure point-to-point or site-to-site connections in routed or bridged

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configurations and remote access facilities. It uses a custom security protocol that utilizes SSL/TLS

for key exchange. It is capable of traversing network address translators (NATs) and firewalls. It

was written by James Yonan and is published under the GNU General Public License (GPL).

OpenVPN allows peers to authenticate each other using a pre-shared secret key, certificates, or

username/password. When used in a multiclient-server configuration, it allows the server to

release an authentication certificate for every client, using signature and Certificate authority. It

uses the OpenSSL encryption library extensively, as well as the SSLv3/TLSv1 protocol, and

contains many security and control features.

OpenVPN Server

To create your OpenVPN Server, navigate to web menu Network Settings->VPN Server. Check the

radio button of OpenVPN to configure your OpenVPN server.

 Enable: Tick to enable OpenVPN server.

 Stealth: Certain deep packet inspection firewalls might not allow OpenVPN traffic, stealth

SSL tunneling can disguise your OpenVPN traffic under the HTTPS traffic which is often seen

as HTTPS traffic by the DPI.

 Certificate: Certificate is one of the client authentication methods available in OpenVPN.

 Port: OpenVPN service port, the default is 1194.

 Stealth Port: Stealth service port, the default is 443.

 Protocol: You can choose either UDP or TCP. Stealth requires TCP only so if you have stealth

enabled then this option is not configurable and the Server will use TCP by default.

 Device Node: TUN or TAP; A TAP device is a virtual Ethernet adapter, while a TUN device is a

virtual point-to-point IP link.

 Cipher: Cipher (or cypher) is an algorithm for performing encryption or decryption.

 Compress Lzo: LZO is an efficient data compression library which is suitable for data

de-compression in real-time.

 TLS-Server: TLS is an excellent choice for authentication and key exchange mechanism of

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OpenVPN.

 Remote Network: OpenVPN remote network.

 Route: The route entries adjust the local routing table, telling it which network to route over

the VPN.

 Client-to-Client: Client-to-Client can enable intercommunication between clients.

5.3.4 IPSec VPN

Internet Protocol Security (IPsec) is a protocol suite for secure Internet Protocol (IP)

communications by authenticating and encrypting each IP packet of a communication session.

IPSec can be configured to operate in two different modes, Tunnel and Transport mode. Use of

each mode depends on the requirements and implementation of IPSec.

IPSec VPN Server (Tunnel mode)

Tunnel mode is used to encrypt all traffic between secure IPSec Gateways, for example if you

have two Syn_PBXs and each acts as an IPSec Gateway for the hosts/IP phones behind it. The

WAN ports will be used to connect both Syn_PBX systems to establish IPSec VPN connection, now

all PCs or IP phones on the LAN ports can communicate with each other on both sides via a

secure IPSec tunnel.

Navigate to web menu Network Settings->VPN Server. Check the IPSec radio button to configure

IPSec VPN server.

 Enable: Tick the checkbox to enable IPSec VPN server.

 Type: Defaults to Tunnel mode.

 IPSec Local IP:Syn_PBX WAN IP, which can be used to connect to the client network.

 IPSec Password: Define a password for authentication of the IPSec client.

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 IPSec Remote IP: IPSec VPN client IP. The client uses this IP to connect to IPSec server.

 IPSec Remote Network: Specify the IPSec VPN client LAN network address.

Notice:

1. If the Syn_PBX is behind NAT, port 500 and 4500 must be open on the router/firewall.

2. If the Syn_PBX is connected to the Internet via PPPoE, then IPSec Local IP needs to be the IP address assigned by

PPPoE.

3. IPSec VPN server can connect 3 IPSec clients.

IPSec VPN Client (Tunnel mode)

On the remote site, open the web GUI of another Syn_PBX system and navigate to web menu to

configure the VPN Client Network Settings->VPN Client.

On the VPN Client page choose IPSec and tick “Enable” option to enable IPSec client.

 Enable: Tick the checkbox to enable IPSec client.

 Type: Ensure this is the same as the IPSec server.

 IPSec Local IP: WAN port IP which can connect to the IPSec server.

 Server Address: Specify the IPSec server IP.

 IPSec Password: Specify the IPSec VPN password defined previously on the server.

 IPSec Remote Network: The IPSec VPN server LAN network address.

Notice:

1. After saving the configuration, the client will try to connect to the server using the details provided.

2. If connection is successfully established then the system will display “Status: 1 tunnel has been established!!!”

3. If connection fails then the system will display “Status: There’s no tunnel! Reconnecting…”

IPSec VPN server (Transport mode)

IPSec Transport mode is used for end-to-end communications, NAT traversal is not supported

with the transport mode. So if two Syn_PBXs are connected via IPSec transport mode, IPSec only

encrypts the communication service ports, unlike Tunnel mode which encrypts the whole LAN

subnet.

Navigate to web menu Network Settings->VPN Server. Check the IPSec radio button.

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 Enable: Tick the checkbox to enable IPSec VPN server.

 Type: Select Transport mode.

 IPSec Local IP:Syn_PBX WAN IP.(This is the same as configuring in Tunnel mode)

 IPSec Password: Define a password for authentication of the IPSec client.

IPSec VPN Client(Transport mode)

On the remote site, open the client SYN_PBX web GUI and navigate to web menu Network

Settings->VPN Client. Check the radio button of IPSec.

 Enable: Tick the checkbox to enable IPSec VPN client.

 Type: Ensure this is the same as the IPSec VPN server.

 IPSec Local IP:Syn_PBX WAN IP which can connect to the IPSec server.

 Server Address: IPSec VPN server IP.

 IPSec Password: Specify the IPSec VPN password defined previously on the server.

Notice:

If a successful connection is established, then the system will display “Status: 2 tunnels have been established!!!”.

Because theSyn_PBX system encrypts all service ports over UDP and TCP protocols, this means there will be 2

tunnels established.

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5.3.5 N2N VPN Client

N2N is an open source Layer 2 over Layer 3 VPN application which utilizes a peer-to-peer

architecture for network membership and routing.

OnSyn_PBX system we support N2N VPN client, to configure the N2N VPN client, please navigate

to web menu Network Settings->VPN Client. Check the radio button of N2N VPN and configure

the client info.

 Enable: Tick this checkbox to enable N2N VPN client.

 Server Address: N2N server(supernode) IP address.

 Port: N2N service port number. This is 82 by default.

 Local IP: VPN local IP.

 Subnet Mask: Netmask of the VPN network.

 Local Port: N2N local service port.

 Username/Password: Used for the N2N server to authorize the connection.

5.4 DHCP Server

DHCP(Dynamic Host Configuration Protocol)is a standardized network protocol used on Internet

Protocol (IP) networks for dynamically distributing network configuration parameters, such as IP

addresses for interfaces and services.

With DHCP, computers/IP phones request IP addresses and networking parameters automatically

fromSyn_PBX WAN/LAN port which saves administrators a lot of time when compared with

having to configure these settings manually.

5.4.1 DHCP Service

Navigate to web menu Network Settings->DHCP Server.

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 Enable: Enable DHCP service.

 Interface: Choose the network port to implement DHCP service.

 Start IP, End IP: Specify the DHCP IP address pool.

 Subnet Mask: Netmask to be assigned to client devices.

 Gateway: Gateway address to be assigned to client devices.

 Primary DNS:DNS to be assigned to client devices.

 Lease Time(min): Duration for DHCP server to lease an address to a new device. When the

lease expires, the DHCP server might assign the IP address to a different device. Default

value is 1440 minutes.

 TFTP Server: Input the TFTP server address if required which may be used to auto provision

your IP phones.

5.4.2 DHCP Client List

Navigate to Network Settings->DHCP Server->DHCP Client List and you will see a list of all devices

receiving their IP address from theSyn_PBX system.

5.4.3 Static Mac

Static MAC is a useful feature which ensures the DHCP service on Syn_PBX always assigns the

same IP address to a specific computer or IP phone on your LAN. To be more specific, the DHCP

service assigns this static IP to a unique MAC address assigned to each NIC on your LAN.

To create a static Mac, navigate to web menu Network Settings->DHCP Server->Static MAC. Click

“New Static MAC” to add a record to the Syn_PBX system.

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In this example, the IP address 192.168.1.123 will always be assigned to the device with MAC

address 6E:72:C3:D4:E5:F6, lease time will not apply to this IP Address.

5.5 DDNS

Unlike DNS that only works with static IP addresses, DDNS (Dynamic Domain Name Server) is

designed to also support dynamic IP addresses, such as those assigned by a DHCP server.

Built-in DDNS feature onSyn_PBX system only requires you to sign up with a Dynamic DNS

provider, then with the domain name they provide which maps your IP address on the Internet,

you can accessSyn_PBX and also other services within your LAN via the domain name without

needing to know your Dynamic public IP Address.

After setting DDNS,Syn_PBX IP PBX phone services can be accessed from remote site via the

domain name which your DDNS provider supplied you. Also remote management is possible,

even without a static public IP.

Syn_PBX system supports the following DDNS service providers:

 http://dyn.com/

 http://www.noip.com/

 http://www.zoneedit.com/

 http://www.oray.com/

 http://www.3322.net

 http://freedns.afraid.org/

Once you have your account details, navigate to web menu Network Settings ->DDNS Settings.

 Enable: Tick to enable DDNS service.

Synway Syn_PBX80 U20V2, Syn_PBX80 U50V2, Syn_PBX80 U100, Syn_PBX100 U20V2, Syn_PBX100 U50V2 User Manual

Specifications and Main Features

Frequently Asked Questions

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