Yeastar Technology MyPBX-SOHO User manual

SOHO
User Manual
Version 1.0.1.1
Yeastar Technology Co., Ltd

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Table of Contents
1. Introduction............................................................................................4
1.1 Features............................................................................................4
1.2 Hardware Specification........................................................................4
1.2.1 Exterior Appearance...................................................................... 4
1.3 Package Contents ...............................................................................6
2. System set up......................................................................................... 7
2.1 Connection Drawing ............................................................................7
2.2 Connecting Ethernet Line.....................................................................7
2.3 Connecting Power ...............................................................................7
3. Manage MyPBX........................................................................................9
3.1 Manager Log in...................................................................................9
3.2 Status Monitor.................................................................................. 10
3.2.1 Line Status................................................................................. 10
3.3 Basic............................................................................................... 11
3.3.1 Extension................................................................................... 11
3.3.1.1 Analog Extension (FXS)）....................................................... 11
3.3.1.2 VOIP Extension ..................................................................... 13
3.3.2 Trunk ........................................................................................ 16
3.3.2.1 Analog Trunk (FXO)............................................................... 16
3.3.2.2 VOIP Trunk........................................................................... 18
3.3.2.3 Service Provider.................................................................... 19
3.3.3 Outbound Routes ........................................................................ 20
3.3.3.1 Create Outbound Route.......................................................... 21
3.3.3.2 Delete Outbuond Route.......................................................... 22
3.3.4 Auto Provision ............................................................................ 22
3.3.4.1 Create New Phone................................................................. 22
3.4 Inbound Call Control ......................................................................... 24
3.4.1 IVR ........................................................................................... 24
3.4.1.1 Create IVR............................................................................ 24
3.4.1.2 Delete IVR............................................................................ 26
3.4.2 IVR Prompts............................................................................... 26
3.4.3 Ring Groups ............................................................................... 27
3.4.3.1 Create Ring Group................................................................. 27
3.4.3.2 Delete Ring Group ................................................................. 28
3.4.4 Inbound Routes .......................................................................... 28
3.4.4.1 Create Inbound Route............................................................ 29
3.4.4.2 Delete Routes ....................................................................... 31
3.4.5 Blacklist..................................................................................... 32
3.4.5.1 Create Blacklist ..................................................................... 32
3.5 Other Settings.................................................................................. 33

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3.5.1 DISA ......................................................................................... 33
3.5.2 Conferencing .............................................................................. 34
3.5.3 Time Conditions.......................................................................... 35
3.5.4 Feature Codes ............................................................................ 35
3.5.5 Paging and Intercom ................................................................... 38
3.6 System Settings ............................................................................... 40
3.6.1 Options...................................................................................... 40
3.6.2 Music on Hold Prompts................................................................. 41
3.6.3 SIP Settings ............................................................................... 42
3.6.4 Voicemail Settings....................................................................... 44
3.6.5 Network Setting.......................................................................... 46
3.6.6 DHCP Server .............................................................................. 47
3.6.7 Autobackup setting...................................................................... 48
3.6.8 Password Settings....................................................................... 49
3.6.9 Date and Time............................................................................ 50
3.6.10 Backup and Restore................................................................... 51
3.6.11 Reset and Reboot...................................................................... 52
3.6.12 Firmware Update....................................................................... 53
3.7 Reports ........................................................................................... 54
3.7.1 Call Logs.................................................................................... 54
3.7.2 System Info ............................................................................... 55
4. Access MRI........................................................................................... 57
5. Use MyPBX ........................................................................................... 62
5.1 Make outbound call........................................................................... 62
5.1.1 Make call via PSTN trunk.............................................................. 63
5.1.2 Make call via VoIP trunk............................................................... 63
5.2 Incoming call.................................................................................... 66
5.2.1 Sample Routing to an IVR............................................................ 66
APPENDIX A FAQ....................................................................................... 68
APPENDIX B How to configure autobackup.................................................... 69
APPENDIX C How to configure NAT setting ................................................... 71
APPENDIX D How to Use Auto Provision ....................................................... 74

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1. Introduction
MyPBX — IPPBX for Small Office and Home Office
1.1 Features
● Auto-provision ● Call Transfer
● IVR(Automated Attendant) ● Call Waiting
● Blind Transfer ● Caller ID
● Call Detail Records (CDR) Search ● Do Not Disturb(DND)
● Call Forwarding ● Music On Hold
● Call Parking ● Ring Group
●Call Pickup ● Caller ID Routing
● Call Recording ● Three-way Conference Calling
● Inbound/Outbound Call Routes ●MRI(MyPBX Recording Interface)
● Paging and Intercom Groups ●Voicemail
● Direct Inward Subscriber Access
(DISA) ● Blacklist
1.2 Hardware Specification
1.2.1 Exterior Appearance
1) Front Side

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Figure 1-1 MyPBX Front Panel Picture
No. Identifying
①Green Light：Indicates the power connection is normal.
②Green Light：Indicates the server system is in working order
③Green Light：Indicates the system is ready.
④Green Light：Indicates the internet interface is in use
⑤Red Light：stands for FXO port
Green Light：stands for FXS port
LED Dual - Red blink：FXO port isn’t connecting PSTN line.
LED Dual - Red and Green blink：FXO port receive an incoming call.
LED Dual - Red and Green fast blink：FXO port is in talking.
LED Dual - Green and Red blink：FXS port is ringing.
LED Dual - Green and Red blink：FXS port is in talking.
2) Back Side
①②③④⑤
Figure 1-2 MyPBX Back Side Picture

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1.3 Package Contents
▲1 unit of MyPBX-SOHO server
▲1 power supply
▲1 Internet line
▲Few telephone cables [RJ-11]
Note: If any of the above items is damaged or missing, contact your reseller.

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2. System set up
2.1 Connection Drawing
2.2 Connecting Ethernet Line
MyPBX provides two 10/100M Ethernet ports with RJ45 interface and LED
indicator. Plug Ethernet line into MyPBX’s Ethernet port, and then connect
the other end of the Ethernet line with a hub, switch, router, LAN or WAN.
Once connected, check the status of the LED indicator. A yellow LED
indicates the port is in the connection process, and a green LED indicates the
port is properly connected.
2.3 Connecting Power
MyPBX utilizes the high-performance switch power, which supply the
enough voltage and electrical energy that required by MyPBX system.
AC Input: 100~240V
DC Output: 12V,1A
PSTN
PSTN
Normal Phone IPPhone
601
602
FXS Ports LAN
FXO Ports IPProvider
Soft Phone
Router

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Please follow the steps below to connect the MyPBX unit to a power outlet:
1. Connect the small end of the power cable to the power input port on the
MyPBX back panel, and plug the other end of the cable into a 100VAC
power outlet.
2. Check the Power LED on the front panel. A solid green LED indicates that
power is being supplied correctly.

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3. Manage MyPBX
3.1 Manager Log in
From your web browser, input the IP address of the MyPBX server.
If this is the first time you are configuring MyPBX, please use the default
settings below:
IP Address: http://192.168.5.150
Username: admin
Password: password
Figure 3-1

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3.2 Status Monitor
3.2.1 Line Status
Figure 3-2
MyPBX Status Description:
Extensions:
1) : extension is unregistered
2) : extension is idle
3）: extension is ringing
4）: extension is busy in a call
Trunks:
SIP Trunk:
1) Unregistered — trunk registration failed.
2) Registered — succeed registration, trunk is ready to use.
3) Request Send — Registering.

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4）Waiting — Waiting for Authentication.
FXO Trunk:
FXO: please refer to the LED identifying of front panel.
3.3 Basic
3.3.1 Extension
Extension has two types: normal desk extension (FXS) and VOIP extension.
Figure 3-3
3.3.1.1 Analog Extension (FXS)）
Edit Analog Extension
On the administration page of FXS extensions, click ‘Edit’ on the extension that
you want to edit, and modify the following information on prompt window
1) General
·Extension
The numbered extension, i.e. 1234, that will be associated with this particular
User / Phone.
·Name
A character-based name for this user, i.e. ‘Bob Jones’

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·Caller ID
The Caller ID (CID) string will be used when this user calls another internal user.
2) Voice Mail
·Voice Mail Access PIN Code
Voicemail Password for this user, Ex: ‘1234’.
·Send Voice Mail to Email Address
This option defines whether or not voicemails are sent to the Email address as
an attachment.
Note: You need to have an SMTP server configured for this functionality.
3) Flash
·rxflash
Sets the amount of time, in milliseconds, that must have passed since the last
hook-flash event received by MyPBX before it will recognize a second event. If
a second event occurs in less time than defined for Flash, then MyPBX will ignore
the event. The default value of Flash is 1000 ms, and it can be configured in 1ms
increments.
4) Group
·Pickup Group
If a phone is assigned to a pickup group that is shared with a ringing phone, that
phone may pick up the incoming call by calling *4 .*4 is default setting, but it
can be changed under Feature Codes -> General -> Call Pickup.
5) Follow me
The mode of call forwarding when there’s incoming call on this extension. Admin
can set up the call forwarding type for this extension.
6）Other Options
.Call Waiting
Check this option if the User or Phone should have Call-Waiting capability.If this
option is checked, busy transfer will not be available for this extension.
.Hasmanager/Web Access
Check this option to allow user to access the MyPBX Web Interface, which allows
a user to manage their saved Voicemail messages and recordings. Users may
access MRI by login to MyPBX web interface with their extension number and
voicemail password.

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Figure 3-4
3.3.1.2 VOIP Extension
A VOIP extension is an SIP Account that allows an IP Phone or an IP Soft- Phone
client to register on MyPBX.
1. Add VOIP Extension
Go to Extensions VOIP Extensions Create New Extension
1) General
·Extension
The numbered extension, i.e. 1234, that will be associated with this particular
User / Phone.
·Password
The password for the user's sip account , Ex: ‘12t3f6’
·Name
A character-based name for this user, i.e. ‘Bob Jones’
·Caller ID
The Caller ID will be used when this user calls another internal extension.

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2) Voice Mail
·Voice Mail Access PIN Code
Voicemail Password for this user, Ex: ‘1234’.
·Send Voice Mail to Email Address
This option defines whether or not voicemails are sent to an Email Address as
attachment.
Note: You need to have an SMTP server configured for this functionality.
3) VoIP Setting
·NAT
Try this setting when MyPBX is on a public IP and communicating with devices
hidden behind a NAT device (broadband router). If you have one-way audio
problems, you usually have problems with your NAT configuration or your
firewall's support of SIP or RTP ports.
·DTMF Mode – RFC2833, Info, Inband, Auto.
4) IP Restriction
·Enable IP Restriction
Check this option to enhance the VoIP security for MyPBX. If this option is
enabled, only the permitted IP or network will be able to access MyPBX’s VoIP
Server.
·Permitted IP/Network
The certain IP or network permitted to access MyPBX’s VoIP Server.
The input format should be : IP address/network mask
Ex: 192.168.5.100/255.255.255.255 – Will allow traffic from IP 192.168.5.100
Ex: 120.42.46.120/255.255.255.255 – Will allow from IP 20.42.46.120
Ex: 192.168.5.0/255.255.255.0 – Will allow from network 192.168.5.0
5) Group
·Pickup Group
If a phone is assigned to a pickup group that is shared with a ringing phone, that
phone may pick up the incoming call by pressing *4. *4 is the default setting,
but it can be changed under Feature Codes -> General -> Call Pickup.
6) Follow me (Call Forwarding)
This function sets inbound call forwarding on an extension. An administrator can
configure Follow Me for this extension.
7）Other Options
.Call Waiting
Check this option if the User or Phone should have Call-Waiting capability.If this
option is checked, busy transfer will not be available.
.Hasmanager/Web Access

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Check this option to allow a user to access the MRI web interface, which allows
a user to manage their saved voicemail messages and recordings. Users may
access MRI by logging in with their extension number and voicemail password.
Figure 3-5
2. Bulk add SIP Extensions
Go to Extensions SIP Extensions Bulk add Extensions
1）Choose the extension quantity you want to create
2）The initiative extension number when starting

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Figure 3-6
3. Edit VOIP Extension
Click ‘Edit’ on VOIP Extension administration page or click ‘Modify Selected
Extensions’ to edit extension.
3.3.2 Trunk
There are three types of trunks: Analog trunks (FXO), VOIP and Service
Provider trunks.
Figure 3-7
3.3.2.1 Analog Trunk (FXO)
1. Edit Analog Trunk (FXO)
On the analog Trunk administration page, click ‘Edit’ on selected trunk and
modify information on prompt window:

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1）General
1）General
·Trunk Name
A unique label used to identify this trunk when listed in outbound rules,
incoming rules, etc. Ex: 'Port 5'
·Audio Setting
Normally you should not have to adjust your analog ports beyond the initial
calibration. Should you still need to fine tune your audio settings, please make
adjustments as necessary.
2）Advanced Options
·Busy Detection
Busy Detection is used to detect far end hang-up or for detecting a busy signal.
Select “Yes” to turn this feature on.
·Busy Count
If Busy Detection is enabled, it is also possible to specify how many busy tones
to wait for before disconnecting. The default is 4, but better results can be
achieved if set to 6 or even 8. Remember, the higher the number, the more time
required to release a channel. A higher setting lowers the probability that you
will encounter random hang-ups
·Busy Pattern
If Busy Detection is enabled, it is also possible to specify the cadence of your
busy signal. In many countries, it is 500msec on, 500msec off. Without Busy
Pattern specified, MyPBX will accept any regular sound-silence pattern that
repeats <Busy Count> times as a busy signal. If you specify Busy Pattern, then
MyPBX will further check the length of the tone and silence, which will further
reduce the chance of a false positive disconnect.
·Caller ID Start
This option allows you to define the start of a Caller ID signal:
Ring: Start when a ring is received (Caller ID Signaling: Bell_USA, DTMF).
Polarity: Start when a polarity reversal is started (Caller ID Signaling:
V23_UK,V23_JP,DTMF).
Before Ring: Start before a ring is received (Caller ID Signaling: DTMF).
·Caller ID Signaling
This option defines the type of Caller ID signaling to use. It can be set to one of
the following:
Bell: bell202 as used in the United States
v23_UK: Used in the UK
v23_JP: Used in Japan
DTMF: Used in Denmark, Sweden, and Holland

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Figure 3-8
3.3.2.2 VOIP Trunk
1. Add SIP Trunk
Input correct SIP information (provided by VOIP provider). Inaccurate
information will prevent the trunk from registering.
·Provider Name
A unique label to help you identify this trunk when listed in outbound rules,
incoming rules etc. Ex: 'yeastar'.
·Hostname/IP
Service provider’s hostname or IP address.5060 is the standard port number
used by SIP protocol. Don’t change this part if it is not required.
.Domain
VoIP provider’s server domain name.
·Username
Username of SIP account. Used for SIP trunk registration.
.Authorization name
Used for SIP authentication. Leave this blank if not required.
·Password
Password of SIP account.
.Online number
Define the online number that expected by Skype for SIP and some others SIP
Service providers. Leave this field blank if it is no required.
·Enable Outbound Proxy Server

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Figure 3-9
3.3.2.3 Service Provider
1. Add Service Provider
To Create the Service provider definition you need to complete the following
VoIP fields.
·Type
SIP or IAX
SIP – Identifies whether the trunk sends and receives calls by using the VoIP
protocol SIP.
IAX - Identifies whether the trunk sends and receives calls by using the VoIP
protocol IAX.
·Provider Name
A unique label would help to you identify this trunk. Ex: 'Provider2'.
·Hostname/IP
Service provider’s hostname or IP address.
Note: 5060 is the standard port number used by SIP protocol, 4569 is the
standard port number used by IAX protocol. Don’t change this part if it is not
required.
.DOD Setting
1) DOD – Direct Outward Dialing Number.
2) Bind Extension.

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Figure 3-10
3.3.3 Outbound Routes
Outbound routing mainly works for guides outgoing calls to go through trunks.
Figure 3-11