Xiamen Yeastar Information Technology STD IP PBX User Manual MyPBX UserManual en

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User Manual
Version 2.16.0.58
Yeastar Technology Co., Ltd.
MyPBX User Manual
Table of Contents
1. Introduction .......................................................................................... 4
1.1 Features ........................................................................................... 6
1.2 Hardware Specifications ...................................................................... 7
1.2.1 Exterior Appearance ..................................................................... 7
2. System set up........................................................................................ 8
2.1 Connection Drawing ........................................................................... 8
2.2 Connecting Ethernet Line .................................................................... 8
2.3 Supplying Power ................................................................................ 9
3. Managing MyPBX .................................................................................. 10
3.1 Administrator Login .......................................................................... 10
3.2 Status Monitor................................................................................. 11
3.2.1 Line Status ................................................................................ 11
3.3 Basic.............................................................................................. 12
3.3.1 Extension .................................................................................. 12
3.3.2 Trunk ....................................................................................... 21
3.3.3 Outbound Routes ....................................................................... 35
3.3.4 Phone Provisioning ..................................................................... 38
3.4 Inbound Call Control ........................................................................ 46
3.4.1 IVR .......................................................................................... 46
3.4.2 Queues ..................................................................................... 49
3.4.3 Custom Prompts ........................................................................ 53
3.4.4 Ring Groups .............................................................................. 53
3.4.5 Inbound Routes ......................................................................... 55
3.4.6 Blacklist .................................................................................... 60
3.5 Internal Settings.............................................................................. 62
3.5.1 Options..................................................................................... 62
3.5.2Business Hours ........................................................................... 64
3.5.3 Feature Codes ........................................................................... 65
3.5.4 SIP Settings .............................................................................. 68
3.5.5 IAX Settings .............................................................................. 73
3.5.6 Voicemail Settings ...................................................................... 75
3.5.7 SMS Settings ............................................................................. 77
3.5.8 DISA ........................................................................................ 80
3.5.9 Conferencing ............................................................................. 81
3.5.10 Paging Groups ......................................................................... 82
3.5.11 DNIS Settings .......................................................................... 84
3.5.12 PIN User Settings ..................................................................... 84
3.5.13 Callback Settings ...................................................................... 85
3.5.14 Speed Dial Settings .................................................................. 87
3.5.15 Music on Hold Prompts .............................................................. 88
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3.6 Network Settings ............................................................................. 88
3.6.1 LAN Settings ............................................................................. 88
3.6.2 WAN Settings ............................................................................ 90
3.6.3 Static Route .............................................................................. 91
3.6.4 Firewall..................................................................................... 92
3.6.5 DHCP Server ............................................................................. 96
3.6.6 VLAN Settings............................................................................ 96
3.6.7 VPN Settings ............................................................................. 98
3.6.8 DDNS Settings ........................................................................... 99
3.7 System Settings ............................................................................ 100
3.7.1 External Storage ...................................................................... 100
3.7.2 Password Settings .................................................................... 100
3.7.3 System Prompts ...................................................................... 101
3.7.4 Date and Time ......................................................................... 102
3.7.5 Backup and Restore.................................................................. 103
3.7.6 Reset and Reboot ..................................................................... 103
3.7.7 Firmware Update...................................................................... 104
3.8 Reports ........................................................................................ 105
3.8.1 Call Logs ................................................................................. 105
3.8.2 System Logs............................................................................ 106
3.8.3 System Info ............................................................................ 106
4. Access MRI ........................................................................................ 108
5. Use MyPBX ........................................................................................ 112
5.1 Make outbound call ........................................................................ 112
5.1.1 Sample Routing via PSTN Trunk ................................................. 113
5.1.2 Sample Routing via VoIP Trunk .................................................. 114
5.2 Incoming call ................................................................................ 115
5.2.1 Sample Routing to an IVR ......................................................... 115
APPENDIX A FAQ.................................................................................... 118
APPENDIX B How to Configure Autobackup ................................................ 119
APPENDIX C How to Configure NAT setting ................................................ 121
APPENDIX D How to Use Auto Provision..................................................... 123
APPENDIX E How Do I Configure Distinctive Ring Tones ............................... 126
APPENDIX F How to Use Email to SMS....................................................... 128
APPENDIX G How to Use DID ................................................................... 130
APPENDIX H How to Use BLF Key to Choose the PSTN line. .......................... 134
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MyPBX User Manual
1. Introduction
MyPBX
—IP-PBX for Small Businesses/Home Office
MyPBX is a standalone embedded hybrid PBX for small businesses and remote
branch offices of larger organizations (1-100 users per site). MyPBX also offers
a hybrid solution (a combination of VoIP applications using legacy telecom
equipment) alternative for enterprises who are not yet ready to migrate to a
complete VoIP solution.
Caveat
This device complies with part 15 of the FCC Rules. Operation is subject to the
following two conditions: (1) This device may not cause harmful interference,
and (2) this device must accept any interference received, including
interference that may cause undesired operation. Any Changes or modifications
not expressly approved by the party responsible for compliance could void the
user‟s authority to operate the equipment.
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Customer Information
1. This equipment complies with Part 68 of the FCC rules and the requirements
adopted by the ACTA. On the bottom of this equipment is a label that
contains, among other information, a product identifier in the format [US:
X8PIS00BSTD]. If requested, this number must be provided to the
telephone company.
2. A plug and jack used to connect this equipment to the premises wiring and
telephone network must comply with the applicable FCC Part 68 rules and
requirements adopted by the ACTA. A compliant telephone cord and modular
plug is provided with this product. It is designed to be connected to a
compatible modular jack that is also compliant. See installation instructions
for details.
3. If this equipment [Yeastar MyPBX] causes harm to the telephone network,
the telephone company will notify you in advance that temporary
discontinuance of service may be required. But if advance notice isn't
practical, the telephone company will notify the customer as soon as
possible. Also, you will be advised of your right to file a complaint with the
FCC if you believe it is necessary.
4. The telephone company may make changes in its facilities, equipment,
operations or procedures that could affect the operation of the equipment. If
this happens the telephone company will provide advance notice in order for
you to make necessary modifications to maintain uninterrupted service.
5. If trouble is experienced with this equipment [US: X8PIS00BSTD], for repair
or warranty information, Service can be facilitated through our office at:
U.S. Agent Company name:
Commlogik Corporation
Address: 1921 NW 82nd Ave Miami - FL - 33126, USA
Tel: +1 (305) 677-7888
Fax: +1 (305) 677-7889
If the equipment is causing harm to the telephone network, the telephone
company may request that you disconnect the equipment until the problem
is resolved.
6. Please follow instructions for repairing if any (e.g. battery replacement
section); otherwise do not alternate or repair any parts of device except
specified.
7. Connection to party line service is subject to state tariffs. Contact the state
public utility commission, public service commission or corporation
commission for information.
8. If your home has specially wired alarm equipment connected to the
telephone line, ensure the installation of this [US: X8PIS00BSTD] does not
disable your alarm equipment. If you have questions about what will disable
alarm equipment, consult your telephone company or a qualified installer.
9. If the telephone company requests information on what equipment is
connected to their lines, inform them of:
a) The ringer equivalence number [0.0B]
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b) The USOC jack required [RJ11C]
c) Facility Interface Codes (“FIC”) [METALLIC]
d) Service Order Codes (“SOC”) [9.0y]
e) The FCC Registration Number [US: X8PIS00BSTD]
10. The REN is used to determine the number of devices that may be connected
to a telephone line. Excessive RENs on a telephone line may result in the
devices not ringing in response to an incoming call. In most but not all areas,
the sum of RENs should not exceed five (5.0). To be certain of the number of
devices that may be connected to a line, as determined by the total RENs,
contact the local telephone company. The REN for this product is part of the
product identifier that has the format US: AAAEQ##TXXXX. The digits
represented by ## are the REN without a decimal point. For this product the
FCC Registration number is [US:X8PIS00BSTD] indicates the REN would be
0.0B.
Applications
1.1 Features
● Auto-provision
● Blind Transfer
● Firewalls
● Follow me
● BLF Support
● Interactive Voice Response (IVR)
● Blacklist
● Intercom / Zone Intercom
● Call Detail Records(CDR)
● Music On Hold
● Call Forward
● Music On Transfer
● Call Parking
● Paging / Zone Paging
● Call Recording
● PIN Users
● Call Pickup
● Queue
● Call Routing
● QOS
● Call Transfer
● Ring Group
● Call Waiting
● Route by Caller ID
● Caller ID
● Skype Integration (Skype Connect)
● Call Back
● Three-way Calling
● Conference
● Mobility Extension
● SMS to Mail
● External Storage
● Mail to SMS
● DDNS
● Speed Dial
● OpenVPN
● Define Office Time
● T.38
● Direct Inward System Access(DISA)
● Voicemail
● DIDs
● VLAN
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● Distinctive Ringtone
● WAN
● Do Not Disturb(DND)
● PPPoE
● Dial by Name
● Static Route
Note: the features with asterisk (*) will be supported in next version.
1.2 Hardware Specifications
1.2.1 Exterior Appearance
Front Side
①②
③④
⑤
Figure 1-1 MyPBX Front Panel Picture
No.
①
②
③
④
⑤
Identifying
Green LED: Indicates correct power is being supplied to the unit
Green LED: Indicates the MyPBX is fully functional.
Green LED: Indicates stable WAN Port connection
Green LED: Indicates stable LAN Port connection
Red LED: Indicates presence of an FXO/GSM port.
Orange LED: Indicates presence of a BRI port.
Green LED: Indicates presence of an FXS port.
LED Blinking- Red blinking: No connection between FXO port and
PSTN
LED Alternating - Red and Green (slow blink): FXO port is receiving
an incoming call.
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LED Alternating - Red and Green (fast blink): FXO port is in use
LED Dual - Green and Red (slow blink): FXS port is ringing
LED Dual - Green and Red (fast blink): FXS port is in use
2. System set up
2.1 Connection Drawing
A. MyPBX
GSM
PSTN
VoIP Provider
FXS
LAN
Route
601
602
603
604
Normal
IP
Soft
Figure 2-1
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.
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2.3 Supplying Power
MyPBX utilizes the high-performance switch power supply, which supplies
the required power for the unit.
AC Input: 100~240V
DC Output: 12V, 5A
Please follow the steps below to connect 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 100V AC
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. Managing MyPBX
3.1 Administrator Login
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 (your PC should be in the same local network with MyPBX):
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 3.2.1
MyPBX Status Description:
Extensions:
1)
: Extension is unregistered
2)
: Extension is idle
3）
: Extension is ringing
4）
: Extension is busy in a call
5）
: Extension is on hold
Trunks:
VOIP Trunk:
Status
Unregistered: Trunk registration failed.
Registered: Succeed registration, trunk is ready for use.
Request Send: Registering.
Waiting: Waiting for authentication.
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FXO Trunk:
Status
Idle: The port is idle.
Busy: The port is in use.
Disconnected: The port hasn’t connected the PSTN line.
More detail message, please refer to the LED identifying of front panel.
GSM Trunk:
Status
Idle: The port is idle.
Busy: The port is in use.
Signal
: No signal.
: Poor.
: Average.
: Good.
: Excellent.
BRI Trunk:
Status
Ok: The ports connect correctly.
Disconnected: The port hasn’t connected the PSTN line.
Service Provider:
Status
OK: Succeed registration, trunk is ready for use.
Unreachable: The trunk is unreachable.
Failed: Trunk registration failed.
3.3 Basic
3.3.1 Extension
Extension has two types: Analog extensions (FXS) and VOIP extensions (SIP
extension or IAX extension).
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Figure 3.3.1
3.3.1.1 Analog Extensions (FXS)
Edit Analog Extensions
On the administration page of FXS extensions, click ‘Edit’ on the extension that
you want to edit, and modify the following information on the popup window
1) General
·Extension
The numbered extension, i.e. 1234, that will be associated with this particular
User / Phone.
·Port
The extension correspond port.
·Name
A character-based name for this user, i.e. ‘Bob Jones’ .
·Caller ID
The Caller ID (CID) string will be used when this user calls another internal user.
2) Voicemail
·Enable Voicemail
Check this box if the user should have a voicemail account.
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·Voicemail Access PIN #
Voicemail Password for this extension, i.e. ‘1234’ .
3) Mail Setting
·Enable Send Voicemail
Once enabled, the voicemail will be sent to the below email address as an
attachment.
·Send Voicemail to Email Address
This option defines whether or not voicemails/Fax is sent to the Email address
as an attachment.
Note: Please ensure that all voicemail settings are properly configured on the
System Settings -> Voicemail Settings page before using this feature.
4) Flash
·Hook Flash Detection
Sets the amount of time, in milliseconds, that must pass 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 by Hook Flash Detection, then
MyPBX will ignore the event. The default value of Flash is 1000 ms, and it can be
configured in 1ms increments.
5) Group
·Pickup Group
If this extension belongs to a pickup group, any calls that ring this extension can
be picked up by other extensions in the same pickup group by dialing the Call
Pickup feature code (default *4).
Note: *4 is the default setting, 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 extension should have Call Waiting capability. If this
option is checked, the ‘When busy’ follow me options will not be available.
·User Web Interface
Check this option to allow the user to login to the MyPBX User Web interface,
which can be used to access voicemail and extension recordings. Users may
login to the MyPBX User Web interface by using their extension number and
voicemail PIN # as the login and password respectively.
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·DND
Don’t Disturb.
·Ring Out
Check this option if you want to custom the ring time. Tone will stop over the
time defined.
8) Volume Settings
Rxgain: The Volume sent to FXS extension.
Txgain: The Volume sent out by the FXS extension.
9) Mobility Extension
MyPBX allows you to use your mobile phone as extension. If you set your mobile
phone as mobility extension and then you call MyPBX with this mobility phone,
you will hear a dial tone. MyPBX will recognize your call as a call from an
extension. You can dial the number of other extensions (Your caller ID will be
the number of your extension) and use all outbound route that your extension
can use of MyPBX.
·Mobility Extension Number
Don't forget to add the dial patterns of the outbound route at the beginning of
your mobile phone number when you fill in the mobility extension number filed.
E.g. if you want to set “15960XXXXXX” as mobile extension, and the dial pattern
of the outbound route is “9”; you should set “915960XXXXXX” here.
Note: If callback is enabled in the inbound route, the mobility extension function
of this inbound route will be disabled.
10) Spy Settings
MyPBX allows extension to monitor/barge in other conversation. Once this
feature is enabled, the extension has the ability to monitor/barge in other calls
using the feature codes for each spy mode, refer to 'Feature Codes' page for
more information.
·spy modes
There are 3 spy modes available for choice:
Normal spy: you can only hear the call, but can't talk
Whisper spy: you can hear the call, and can talk with the monitored extension
Barge spy: you can hear the call and talk with them both
Note: for example, if 500 want to monitor extension 501, we need to enable the
'allow being spied ' for 501, and choose the spy mode for extension 500.
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Then pick up 500 and dial “feature codes + 501” to start monitoring when 501 is in a
call.
Figure 3.3.1.1
3.3.1.2 VOIP Extension
A VOIP extension is a SIP/IAX Account that allows an IP Phone or an IP SoftPhone client to register on MyPBX.
1. Add VOIP Extension
Go to Extensions  VOIP Extensions  Create New Extension
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1) General
·Type
Extension type: SIP,IAX or SIP/IAX.
SIP – The extension sends and receives calls using the VoIP protocol SIP.
IAX -The extension sends and receives calls using the VoIP protocol IAX.
·Extension
The numbered extension, i.e. 1234, that will be associated with this particular
User / Phone.
·Password
The password for this extension, Ex: ‘12t3f6’
·Name
A character-based name for this user, EX: ‘Bob Jones’
·Caller ID
The Caller ID will be used when this user calls another internal extension.
2) Voicemail
·Enable Voicemail
Check this box if the user should have a voicemail account.
·Voicemail Access PIN #
The voicemail Password for this extension, i.e. ‘1234’ .
3) Mail Setting
This option defines whether or not voicemails or faxes are sent to an Email
Address as attachment.
·Enable Send Voicemail
Once enabled, the voicemail will be sent to email as an attachment.
·Email Address
Email address used to receive the voicemail or Fax.
Note: Please ensure that the section 'SMTP Settings For Voicemail'(in the
'Voicemail Settings') have been properly configured before using this feature.
4) Group
·Pickup Group
If this extension belongs to a pickup group, any calls that ring this extension can
be picked up by other extensions in the same pickup group by dialing the Call
Pickup feature code (default is *4).
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Note: *4 is the default setting, it can be changed under Feature Codes ->
General -> Call Pickup.
5) Follow me (Call Forwarding)
Call forwarding for an extension can be configured here. The administrator can
configure Follow Me option for this extension. If you want to transfer the call to
an outbound number, please follow the dial pattern of outbound route filled in
the outbound number.
For example: transferring to your mobile phone number 123456789, the dial
pattern of outbound route is ‘9.’, you should fill in 9123456789 here.
6）Other Options
.Call Waiting
Check this option if the extension should have Call Waiting capability. If this
option is checked, the ‘When busy’ follow me options will not be available. The
call waiting function of IP phone has higher priority than MyPBX’s call waiting
function.
.DND
Don’t Disturb. When DND is enabled for an extension, the extension will be not
available.
.User Web Interface
Check this option to allow the user to login to the MyPBX User Web interface,
which can be used to check voicemail and extension recordings. Users may login
to MyPBX User Web interface by using their extension number and voicemail PIN
# as the login and password respectively.
.Ring Out
Check this option if you want to customize the ring time. Ring tone will stop over
the time defined.
7) VoIP Settings
·NAT
This setting should be used when the system is using a public IP address to
communicate with devices hidden behind a NAT device (such as a broadband
router). If you have one-way audio problems, you usually have problems with
your NAT configuration or your firewall's support of SIP and/or RTP ports.
·Enable SRTP
Enable extension for SRTP (RTP Encryption).
·Qualify
Send check alive packets to IP phones
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·SIP Transport
This will be the transport method used by the extension. The options are UDP
(default) or TCP or TLS.
·DTMF Mode – RFC2833, Info, Inband, Auto.
8) IP Restriction
·Enable IP Restriction
Check this option to enhance the VoIP security for MyPBX. If this option is
enabled, only the permitted IP/Subnet mask will be able to register this
extension number. In this way, the VoIP security will be enhanced.
·Permitted ‘IP address/Subnet mask’
The input format should be ‘IP address’+’/’+’Subnet mask’.
e.g."192.168.5.100/255.255.255.255" means only the device whose IP
address is 192.168.5.100 is allowed to register this extension number.
e.g."192.168.5.0/255.255.255.0" means only the device whose IP address is
192.168.5.XXX is allowed to register this extension number.
9) Mobility Extension
MyPBX allows you to use your mobile phone as extension. If you set your mobile
phone as mobility extension and then you call MyPBX with this mobility phone,
you will hear a dial tone. MyPBX will recognize your call as a call from an
extension. You can dial other extension numbers (Your caller ID will be the
number of your extension) and dial out using all outbound route that your
extension can use on MyPBX.
·Mobility Extension Number
Don't forget to add the dial patterns of the outbound route at the beginning of
your mobile phone number when you fill in the mobility extension number filed.
E.g. if you want to set “15960XXXXXX” as mobile extension, and the dial pattern
of the outbound route is “9”; you should set “915960XXXXXX” here.
Note: If callback is enabled on the inbound route, the mobility extension
function of this inbound route will be disabled.
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Figure 3.3.1.2
2. Add Multiple Extensions
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Go to Extensions  VOIP Extensions  Add Multiple Extensions.
1) Select the number of extensions that you would like to create.
2) Select the type of extension that you would like to create.
3) Fill in the starting extension number.
Figure 3.3.1.3
3. Edit VOIP Extension
Click ‘Edit’ on VOIP Extension administration page or click ‘Modify Selected
Extensions’ to edit extensions.
3.3.2 Trunk
There are five types of trunks: Analog trunks (FXO), GSM Trunk, BRI Trunk,
VoIP Trunk and Service Provider trunk.
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Figure 3.3.2
3.3.2.1 Analog Trunk (FXO)
1. Edit Analog Trunk (FXO)
On the Trunk administration page, click ‘Edit’ on the selected trunk and modify
its properties in the popup window:
1）General
·Trunk Name
A unique label used to identify this trunk when listed in outbound rules,
incoming rules, etc. Ex: 'pstn5'
·Volume Setting
Used to modify the volume level of this trunk . Normally, this setting does not
need to be changed.
2）Busy Detection
·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.
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·Busy Count
If Busy Detection is enabled, it is also possible to specify how many busy tones
to wait for before disconnecting the call. The default is 4, but better results can
be achieved if set to 6 or even 8. Remember, the higher the number, the more
time will be required to release a channel. A higher setting lowers the
probability that you will encounter random hang-ups.
·Busy Interval
The busy detection interval
·Busy Pattern
If Busy Detection is enabled, it is also possible to specify the cadence of your
busy signal. In many Countries, it is 500 msec on, 500 msec off. Without Busy
Pattern specified, MyPBX will accept any regular sound-silence pattern that
repeats  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.
·Frequency Detection
Used for Frequency Detection (Enable detecting the busy signal frequency or
not).
·Busy Frequency
If the Frequency Detection is enabled, you must specify the local frequency.
·Polarity Detection
Configure if the call needs to be hung up when a polarity signal arrived
3）Advanced Options
·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: suitable in the UK
v23_Japan: suitable in Japan
v23-Japan pure: suitable in Japan
DTMF: suitable in Denmark, Sweden, and Holland
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.Caller ID Detection
For fxo trunks, this option forces MyPBX to clarify Caller ID incoming calls.
Figure 3.3.2.1
3.3.2.2 GSM Trunk
1. Edit GSM Trunk
1）General
·Trunk Name
A unique label used to identify this trunk when listed in outbound rules,
incoming rules, etc. Ex: 'GSM9'
·Volume Setting
Used to modify the volume level of this trunk. Normally, this setting does not
need to be changed.
·Pin Code
Please enter your SIM card pin code here if your card has a pin code.
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Figure 3.3.2.2
3.3.2.3 BRI Trunk
1. Edit BRI Trunk
1）General
·Trunk Name
A unique label used to identify this trunk when listed in outbound rules,
incoming rules, etc. Ex: 'BriTrunk1'
·Signaling
Signaling method.
BRI-CPE: ISDN BRI in TE mode and Point to Point.
BRI-CPE-PTMP: ISDN BRI in TE mode and Point to multi Point.
BRI-NET: ISDN BRI in NET mode and Point to Point.
BRI-NET-PTMP: ISDN BRI in NET mode and Point to multi Point.
·Switch Type
National: National ISDN type2 (common in the US)
ni1: National ISDN type 1
dms100: Nortel DMS100
4ess: AT&T 4ESS
5ess: Lucent 5ESS
euroisdn: EuroISDN
qsig: D-channel signaling protocol at Q reference point for PBX networking.
·PRI Dial Plan
Sets an option required for some (rare) switches that require a dial plan
parameter to be passed. This option is ignored by most BRI switches. It may be
necessary on a few pieces of hardware. This option can almost always be left
unchanged from the default.
·Reset interval
Sets the time in seconds between restart of unused channels . Some PBXs don't
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like channel restarts. so set the interval to a very long interval e.g. 100000000
or 'never' to disable *entirely*. If you are in Israel, the following is important:
As Bezeq in Israel doesn't like the B-Channel resets happening on the lines, it is
best to set the reset interval to 'never' when installing a box in Israel. Our past
experience also shows that this parameter may also cause issues on local
switches in the UK and China.
·PRI Local Dial Plan
Sets an option required for some (rare) switches that require a dial plan
parameter to be passed. This option is ignored by most BRI switches. It may be
necessary on a few pieces of hardware. This option can almost always be left
unchanged from the default.
·Over Lap Dial
Whether MyPBX can dial this switch using overlap digits . If you need Direct
Dial-in (DDI; in German "Durchwahl") you should change this to yes, then
MyPBX will wait after the last digit it receives.
·PRI Indication
Tells how Device should indicate Busy() and Congestion() to the switch/user.
Accepted values are:
inband: Device plays indication tones without answering; not available on all
PRI/BRI subscription lines .
outofband: Device disconnects with busy/congestion information code so the
switch will play the indication tones to the caller. Busy() will now do same as
setting PRI_CAUSE=17 and Hangup().
·Enable Facility
To enable transmission of facility-based ISDN supplementary services (such as
caller name from CPE over facility) .
·NSF
Used with AT&T PRIs. If outbound calls are being rejected due to "Mandatory
information element missing" and the missing IE is 0x20, then you need this
setting.
·Echo Cancellation
Echocancel Obviously this disables or enables echo cancellation, it is
recommended to not turn this off.
·Hide CallerID
If you want others to see your CID, please disable this option.
·Codec
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You can choose alaw or ulaw codes.
2) CallerID Prefix
·International Prefix
When there are international calls coming in via this BRI trunk, the International
Prefix you have set here will be added before the CID. So you can know this is
an international call before you answer it.
·National Prefix
When there are national calls coming in via this BRI trunk, the National Prefix
you have set here will be added before the CID. So you can know this is a
national call before you answer it.
·Local Prefix
When there are Local calls coming in via this BRI trunk, the Local Prefix you have
set here will be added before the CID. So you can know this is a local call before
you answer it.
·Private Prefix
When there are Private calls coming in via this BRI trunk, the Private Prefix you
have set here will be added before the CID. So you can know this is a Private call
before you answer it.
·Unknown Prefix
When there are calls with unknown number coming via this BRI trunk, the
Unknown Prefix you set here will be shown as the caller ID.
3) DOD Setting
·Global DOD
Global direct outward dialing number.
·DOD
Direct Outward Dialing Number.
·Associated Extension
The extension make call out via BRI Trunk will display the associated DOD.
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Figure 3.3.2.3
3.3.2.4 VOIP Trunk
1. Add SIP Trunk
Input correct SIP information (provide by VOIP provider). Inaccurate
information will prevent the trunk from registering.
1) General setting
·Type
SIP – Identifies whether the trunk sends and receives calls using the VoIP
protocol SIP
·Provider Name
A unique label to help you identify this trunk when listed in outbound rules,
incoming rules etc. Ex: 'yeastar'.
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·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 .
.From User
All outgoing calls from this SIP Trunk will use the From User (In this case the
account name for SIP Registration) in From Header of the SIP Invite.
.Online number
Define the online number that expected by ‘Skype Connect’ and some other SIP
service providers. Leave this field blank if it's no required.
·Maximum Channels
Controls the maximum number of outbound channels (simultaneous calls) that
can be used on this trunk . Inbound calls are not counted against the maximum.
Set as 0 to specify no maximum.
·Caller ID
Specify the caller ID to use when making outbound calls over this trunk. The
caller ID set in the ‘extension’ screen will override the caller ID set in the ‘VOIP
trunk’ screen. Please note that not all the service providers support this feature.
Contact your service provider for more information.
·Outbound Proxy Server
A proxy that receives requests from a client , even though it may not be the
server resolved by the Request-URI.
·Codecs
Define the codec for this sip trunk and its priority
Note: codec can only display when it is edited after creating the trunk.
·Transport
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This will be the transport method used by the SIP Trunk. This method is given by
the SIP trunk provider. The options are UDP (default) or TCP or TLS.
·Enable SRTP
Define if SRTP is enabled for this trunk
·Qualify
Send check alive packets to the sip provider.
·DTMF mode
Set default mode for sending DTMF of this trunk. Default setting: rfc2833
2) DOD Setting
·DOD
Direct Outward Dialing Number.
·Associated Extension
The extension make call out via SIP Trunk will display the associated DOD.
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Figure 3.3.2.4.1
2. Add IAX Trunk
Input correct IAX information (provided by VOIP provider). Inaccurate
information will prevent the trunk from registering.
1) General
·Type
IAX – Identifies whether the trunk sends and receives calls by using the VoIP
protocol IAX.
·Provider Name
A unique label to help you identify this trunk when listed in outbound rules,
incoming rules etc. Ex: 'yeastar2'.
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·Hostname/IP
Service provider’s hostname or IP address. 4569 is the standard port number
used by IAX protocol. Don’t change this part if it is not required.
·Username
Username of IAX account . Used for IAX trunk registration.
·Password
Password of IAX account .
.Online number
Define the online number that expected by ‘Skype Connect’ and some other SIP
service providers. Leave this field blank if it's no required.
·Maximum Channels
Controls the maximum number of outbound channels (simultaneous calls) that
can be used on this trunk . Inbound calls are not counted against the maximum.
Set as 0 to specify no maximum.
·Caller ID
Specify the caller ID to use when making outbound calls over this trunk. The
caller ID set in the ‘extension’ screen will override the caller ID setting in the
‘VOIP trunk’ screen. Please note that not all the service providers support this
feature. Contact your service provider for more information.
2) DOD Setting
·DOD
Direct Outward Dialing Number.
·Associated Extension
The extension make call out via BRI Trunk will display the associated DOD.
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Figure 3.3.3.4.2
3.3.2.5 Service Provider
1. Add Service Provider
To Create the Service provider definition you need to complete the following
VoIP fields.
1) General
·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.
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·Maximum Channels
Controls the maximum number of outbound channels (simultaneous calls) that
can be used on this trunk . Inbound calls are not counted against the maximum.
Leave blank to specify no maximum.
·Codecs
Define the codec for this sip trunk and its priority
Note: codec can only display when edit it after creating the trunk.
·Transport
This will be the transport method used by the SIP Trunk. This method is given by
the SIP trunk provider. The options are UDP (default) or TCP or TLS.
·Qualify
Send check alive packets to the sip provider.
·DTMF mode
Set default mode for sending DTMF of this trunk. Default setting: rfc2833
2) DOD Setting
·DOD
Direct Outward Dialing Number.
·Associated Extension
The extension make call out via BRI Trunk will display the associated DOD.
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Figure 3.3.2.5
3.3.3 Outbound Routes
Outbound routing defines how outgoing calls are processed through the trunks.
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Figure 3.3.3
3.3.3.1 New Outbound Route
Click ‘New Outbound Route’ and fill in the corresponding information in the
popup window.
1）General
·Route Name
Name of this Outbound Route . Ex: 'Local' or 'Long Distance' etc.
·Dial Pattern
Outbound calls that match this dial pattern will use this outbound route. There
are a number of dial pattern characters that have special meanings:
X : Any Digit from 0-9
Z : Any Digit from 1-9
N : Any Digit from 2-9
[12345-9] : Any digit in the brackets (in this example, 1,2,3,4,5,6,7,8,9)
The ‘.’ Character will match any remaining digits. For example, 9011. will match
any phone number that starts with 9011, excluding 9011 itself.
The ‘!’ will match none remaining digits, and causes the matching process to
complete as soon as it can be determined that no other matches are possible.
Example 1: NXXXXXX will match any 7 digits phone number.
Example 2: 1NXXNXXXXX will match a phone number starting with a 1,
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followed by a 3-digit area code, and then 6 digit number.
·Strip digits from front
Allows the user to specify the number of digits that will be stripped from the
front of the phone number before the call is placed. For example, if users must
press 0 before dialing a phone number, one digit should be stripped from the
dial string before the call is placed.
·Prepend these digits before dialing
These digits will be prepended to the phone number before the call is placed. For
example, if a trunk requires 10 digit dialing, but users are more comfortable
with 7 digit dialing, this field could be used to prepend a 3 digit area code to all
7 digit phone numbers before calls are placed. When using analog trunks, a 'w'
character may also be prepended to provide a slight delay before dialing.
·Password
The route password can be used to protect this route from being accessed
without a password.
·T.38 Support:
Enable T38 fax in this outbound route (Only for SIP Trunk).
·Rrmemory Hunt
Round robin with memory, remembers which trunk was used last time, and then
use the next available trunk to call out.
·Member Extensions
Defines the extensions that will be permitted to use this outbound route .
·Member Trunks
Defines the trunks that can be used for this outbound route .
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Figure 3.3.3.1
3.3.4 Phone Provisioning
The Auto Provision sub menu provides users a method to Auto Provision IP
Phone after the Express Setup process.
Note: Auto Provision functions fully test with these models:
Yealink (T12,T18,T20,T22,T26,T28,T32,T38,VP530,VP-2009)
Snom ( 300,320,360,370)
Polycom (IP 6000,IP 7000,IP 32X,IP33X,IP430,IP450,IP550,IP560,VVX1500)
Cisco (IP7940,IP7960)
Aastra(480i,480i CT,6757i,6757i CT, 6737i)
News:
When provisioning Yealink and Snom IP phone, MyPBX is not needed to
be set as the only DHCP server any more.
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Figure 3.3.4
3.3.4.1 General Settings for Yealink
In this page, you can configure it before provisioning Yealink IP phones,
including the items like general preferences, codecs, remote phone book and
firmware upgrade
Note: if firmware download server is enabled, IP phone will update the firmware
automatically according the version and server you have configured during the
provision process.
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Figure 3.3.4.1
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3.3.4.2 Phone book
You can add your contacts here and provision them to your IP phone.
Figure 3.3.4.2
1) Add Contact
·Type
There are three types: None, VIP and Deny list (Blacklist).
·Group
There are 5 groups: None, Friends, Family, Work, Colleagues list.
·Nick Name
You can set a nick name for this number.
·Favorite
Only works with snom phone.
·Organization
Input the organization of this contact. Only works with snom phone.
·Title
Input the title of this contact. Only works with snom phone.
·Email
Input the email of this contact. Only works with snom phone.
·Birthday
Input the birthday of this contact. Only works with snom phone.
·First Name
Input the first name of this contact. Only works with snom phone.
·Family Name
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Input the family of this contact. Only works with snom phone.
·Office Number
Input the office number here
·Mobile Number
Input the mobile number here
·Home Number
Input the home number here
·Sub Number
Add sub number of this contact. Only works with snom phone.
·Note
Take some note of this contact. Only works with snom phone.
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Figure 3.3.4.2.1
2) Upload Phonebook
You can upload a phonebook before auto provision, which will be provisioned to
the IP phone when using auto provision feature to configure your IP phones. The
format of phonebook should be *.xml.
Note: All the existing phonebooks of the IP phone will be replaced automatically
if the phonebooks are configured in this way.
3.3.4.3 configure phone
Let's take provisioning yealink as an example
Create New Phone have two modes,
Create New phone in webpage and Upload the IP Phone’s configure file.
1. Add new phone via webpage
Click ‘Add Phone’ and fill in the corresponding information in the popup window.
1) General
· Enabled
Choose yes or no to enable or disable this extension
· MAC address
Input the MAC address of IP phone
·Name
Put the name of this Phone here.
·Manufacturer
You can choose the Manufacturer of IP phone
·Phone Type
Choose the model of your phone. Only for snom phone
·Call Waiting
This call feature allows your phone to accept other incoming calls to an
extension already in an active call.
·Key as Send
Configure the key as send, you choose # ,* or disable it
·Auto redial
Enable the auto redial for IP Phone
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·Auto answer
Configure if auto answer is allowed for IP phone
·Phone book
Enable the feature of phone book of IP phone
·Line
Extension: Selected the extension number for IP Phone.
Label: It is shown on the LCD for users to identify the account.
Line Active: You can choose on/off to enable/disable the account respectively.
Figure 3.3.4.3
2) Audio codec
In this section, we can design the allowed codec for IP phone
Figure 3.3.4.4
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3) Line keys settings
Configure the DSS keys/Function Keys
Figure 3.3.4.5
3.3.4.3 Not configured phone
In this section, MyPBX will scan all the supported IP phones and display here, we
can click the 'MAC address' of IP phone and input the corresponding information
in the popup window, like the picture shows below
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Figure 3.3.4.6
3.3.4.4 Upload a file
Click ‘Upload a file’ and choose the configure file of IP phone in the popup
window.
Note: the file format must be .cfg
Please edit the configuration files in advance before uploading.
3.4 Inbound Call Control
3.4.1 IVR
When there’s an inbound call aims at Auto Attendant, MyPBX will play an IVR
recording and route the caller to the requested destination (for example,
‘Welcome to XX company, for sales press 1, for technical support press 2, for
operator press 0, etc’). The system will transfer the call to corresponding
extension according to DTMF digits inputted by the user.
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Figure 3.4.1
3.4.1.1 Create IVR
Click ‘Create New IVR’.
1）General
·Number
MyPBX treats IVR as an extension; you can dial this extension number to reach
the IVR.
·Name
A name for the IVR
·Prompt
The prompt recording that will be played when this IVR is reached.
·Repeat Count
The number of times that the selected IVR prompt will be played.
·Key Timeout
Wait for the user to enter a new extension for a specified number of seconds.
·Allow Dialing of Other Extensions
Allow the caller to dial other extensions other than the ones explicitly defined.
2）Key Press Events
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A list of actions that can be performed depending on the digit dialed by the user .
·Key
The Key pressed when the callers hear the IVR prompt.
·Action
When the callers press the corresponding key, the action MyPBX executes.
No Action: Do nothing
Connect to Extension: Connect the call to an extension.
Connect to Voicemail: Connect the call to the voicemail of an extension
Connect to RingGroup: Connect the call to a ringgroup.
Connect to IVR:Connect the call to an IVR.
Connnect to Conference Room: Connect the call to a conference room.
Connect to DISA:Connect the call to a DISA.
Connect to Queue:Connect the call to a queue.
Connect to Faxes:Connect the call to Faxes of extensions.
Dial by Name:The callers can dial the name of an extension to connect to the
corresponding extension.
Hung up: Hung up the call.
·Destination
Where will MyPBX route the call when the action occurs.
·Time Out
Defines the timeout action . A timeout occurs after the IVR prompt has finished
playing for the number of times specified by the 'Repeat Count' field.
·Invalid
Defines the invalid action . The invalid action is triggered if the user enters a
DTMF digit that is not defined for this IVR.
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Figure 3.4.1.1
3.4.2 Queues
Call Queues give users (i.e. call centers) an efficient means to have their calls
answered in the order they were received to deliver top tier customer service.
Call queues allow calls to be sequenced to one or more agents.
Note: Dial 'Queue number + '*'' to log in or 'Queue number + '**'' to log out the
queue. For example, if the queue number is '680', then agent can dial '680*' to
log in or '680**' to log out.
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1）General
·Queue Name
A name for the Queue.
·Queue Number
Use this number to dial into the queue, or transfer callers to this number to put
them into the queue.
·Queue Password
You can require agents to enter a password before they can login to this queue.
·Queue Agent Timeout
The number of seconds an agent's phone can ring before we consider it a
timeout.
·Queue Max Wait Time
The maximum number of seconds a caller can wait in a queue before being
pulled out. (0 for unlimited).
·Queue Ring Strategy
This option sets the Ringing Strategy for this Queue. The options are
RingAll: Ring All available Agents simultaneously until one answers.
LeastRecent: Ring the Agent which was least recently called.
FewestCalls: Ring the Agent with the fewest completed calls.
Random: Ring a Random Agent.
RRmemory: Round Robin with Memory, Remembers where it left off in the last
ring pass".
2）Agents
This selection shows all users. Selecting a user here makes them a agent of the
current queue.
3) Caller Position Announcements
·Announce Position
Announce position of caller in the queue
·Announce Hold Time
Enabling this option causes MyPBX to announce the hold time to the caller
periodically based on the frequency timer. Either yes or no; hold time will not be
announced if <1 minute.
·Frequency
How often to announce queue position and estimated hold time.
Note: '0 seconds' means disable the announcement
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4) Periodic Announcements
·Prompt
Select a prompt file to play periodically.
·Frequency
How often to announce a prompt to the caller.
5) Events
If a caller presses the key while waiting in the queue, this setting selects which
action should process the key press.
6) Failover-Destination
Defines the failover action. A failover occurs after the user reach the Queue max
wait time.
7) Others
·Music On Hold
Select the 'Music on Hold' Class for this Queue.
·Leave When Empty
This option controls whether callers already on hold are forced out of a queue
that has no agents. There are two options.
Yes: Callers are forced out of a queue when no agents are logged in.
No: Callers will remain in a queue with no agents.
·Join Empty
This option controls whether callers can join a call queue that has no agents.
There are two options,
Yes: Callers can join a call queue with no agents or only unavailable agents
No: Callers cannot join a queue with no agents
The default option is No.
·Agent Announcement
Announcement played to the Agent prior to bridging in the caller.
·Join Announcement
Announcement played to callers once prior to joining the queue.
·Retry
The number of seconds we wait before trying all the phones again.
·Wrap-up time
How many seconds after the completion of a call an Agent will have before the
Queue can ring them with a new call. The default is 30.
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Figure 3.4.2
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3.4.3 Custom Prompts
1. Record new Prompt
The administrator can use this screen to record custom prompts by doing the
following:
1）Click ‘Record New Custom Prompt’
2) Input the desired file name on the popup window and choose an extension to
call for recording (such as 500).
3) Click ‘Record’. The selected extension will ring and you can pick up the phone
to start recording.
Figure 3.4.3.1
2. Upload Prompt
The administrator can also upload prompts by doing the following:
1)）Click ‘Upload Prompt’.
2) Click ‘Browse’ to choose the desired prompt.
3) Click ‘Upload’ to upload the selected prompt.
Figure 3.4.3.2
3.4.4 Ring Groups
Ring groups can be configured to balance the call traffic for multiple users and
give callers a higher level of availability for incoming calls. Multiple ring methods
and voicemail are supported.
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Figure 3.4.4
3.4.4.1 Create Ring Group
Click ‘New Ring Group’ to enter into the Manage Ring Groups page
1）General
·Ring Group Name
This option defines a name for this group, i.e. ‘Sales’. ‘Ring Group Name’ is a
label to help you identify this group in the group list.
·Ring Group Number
This option defines the numbered extension that can be dialed to reach this
group.
·Strategy
This option sets the Ringing Strategy for this Group. The options are as follows:
1. Ring All Simultaneously: Ring all available Extensions simultaneously.
2. Ring Sequentially: Ring each extension in the group one at a time.
·Seconds to ring each member
1. If the strategy is ‘Ring All Simultaneously’, it means set the number of
seconds to ring this group before routing the call according to the ‘Destination if
No Answer’ settings.
2. If the strategy is ‘Ring Sequentially’, it means set the number of seconds to
ring a single extension before moving onto the next one.
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2) Ring Group Members
An extension can be made a member of this ring group by moving it into the
‘Selected’ box.
3) Destination If No Answer
When all members on this group fail to answer the call, system will handle the
call according to the selected destination.
Figure 3.4.4.1
3.4.5 Inbound Routes
Inbound routing processes incoming call traffic to destination extensions during
office hours or outside office hours.
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Figure 3.4.5
3.4.5.1 Create Inbound Route
Click ‘New Inbound Route’ to enter to the Manage Inbound Routes page.
When an incoming call arrives, the system will first check ‘fax detection’, then
‘Holidays’, at last ‘Business Days’.
1）General
·Route Name
A name for this inbound route. Ex: 'pstncallin' etc.
·DID Number
Define the expected DID Number if this trunk passes DID on incoming calls.
Leave this field blank to match calls with any or no DID info. Only service
provider, E1 trunks, BRI trunks or SIP trunks need to be configured with this
setting.
You can also use pattern matching to match a range of numbers. The following
patterns may be used:
X : Any Digit from 0-9
Z : Any Digit from 1-9
N : Any Digit from 2-9
[12345-9] : Any digit in the brackets (in this example, 1,2,3,4,5,6,7,8,9)
The ‘.’ Character will match any remaining digits. For example, 9011. will match
any phone number that starts with 9011, excluding 9011 itself.
The ‘!’ will match none remaining digits, and causes the matching process to
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complete as soon as it can be determined that no other matches are possible.
Example 1: NXXXXXX will match any 7 digits phone number.
Example 2: 1NXXNXXXXX will match a phone number starting with a 1,
followed by a 3-digit area code, and then 6 digit number.
For more information, please refer to Appendix G How to Use DID.
·Extension
Define the extension for DID number. This field is only valid when you use BRI,
SIP, SPS or SPX trunk for this inbound router. You can only input number and
‘-‘in this field, and the format can be xxx or xxx-xxx. The count of the number
must be only one or equal the count of the DID number.
·Caller ID Number
Define the Caller ID Number to be matched on incoming calls. Leave this field
blank to match any or no DID info.
You can also use a pattern match (e.g. 2[345]X) to match a range of numbers.
The following patterns may be used:
X : Any Digit from 0-9
Z : Any Digit from 1-9
N : Any Digit from 2-9
[12345-9] : Any digit in the brackets (in this example, 1,2,3,4,5,6,7,8,9)
The ‘.’ Character will match any remaining digits. For example, 9011. will match
any phone number that starts with 9011, excluding 9011 itself.
The ‘!’ will match none remaining digits, and causes the matching process to
complete as soon as it can be determined that no other matches are possible.
Example 1: NXXXXXX will match any 7 digits phone number.
Example 2: 1NXXNXXXXX will match a phone number starting with a 1,
followed by a 3-digit area code, and then 6 digit number.
·Distinctive Ringtone
MyPBX support mapping to custom ring tone files. For example, if you configure
the distinctive ringing for custom ring tone to ‘Family’, the ring tone will be
played if the phone receives the incoming call.
·Enable Callback
You can enable the callback function of this inbound route. If you want to
configure the callback function, please refer to chapter 3.5.12
How do I configure distinctive ring tones? Please refer to APPENDIX E.
Currently distinctive ringtone can be compatible with Yealink and Snom phone.
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2) Member Trunks
This area allows you to select which trunks will be member trunks for this route.
To make a trunk a member of this route, please move it to the 'Selected' box.
4) Business Days
Define where the calls will be routed during Business Days.
·Office Days
Select one defined business days office days.
·Office Hours Destination
Configure where to route the incoming calls during office hours.
·End Calls
Route the incoming calls to end calls, System will auto hang-up the call.
·Extension
Route the incoming calls to a specific extension.
·Voicemail
Route the incoming calls to extension’s voicemail.
·IVR
Route the incoming calls to a specific IVR.
·Ring Group
Route the incoming calls to a specific Ring Group.
·Conference Room
Route the incoming calls to a specific Conference Room.
·DISA
Route the incoming calls to a specific DISA.
·Queues
Route the incoming calls to a specific Queue.
·Faxes
Route the incoming faxes to a specific extension’s mail address.
Note: This function only supports T.38 faxes.
·Outbound Routes
Route the incoming calls to a specific outbound route.
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This function is mainly used for the connection of two branches.
For example: Company A locates headquarters in the USA with a branch B in
China. A and B both have MyPBX phone systems .
Now if staff of A would like to make a call to a telephone or mobile phone in
China from the extension of A but via the FXS line of B, that can be done by this
configuration.
·Non-office Hours Destination
Configure where to route the incoming calls during non-office hours.
5) During Holidays
Define where the calls will be routed during Holidays.
·Holiday
Select the which defined Holiday to use.When a time is defined in both Business
Days and Holidays, it will be treated as Holidays.
·Destination
Configure where to route the incoming calls during holidays.
6) Fax Detection
Configure if detecting faxes in this inbound route.
·Destination
Configure where the faxes will be routed when faxes are detected.
·No detect
Do not detect faxes.
·Custom Email
Customize an E-mail address to receive the faxes. You should first configure the
‘Voicemail Settigns->SMTP Settings for Voicemail’ correctly before you use this
option.
·Faxes
Send faxes to an extension. If choosing a FXS extension here, the fax will be
sent to the FXS port selected, you should connect a fax machine to this FXS port.
If Choosing a VoIP extension, the fax will be sent to the extension’s voicemail as
an attachment.
Note: If you receive faxes with custom Email address, the ‘SMTP settings’ of
‘Voicemail Settings’ should be configured successfully in advance. If you receive
faxes with E-mail address configured in VOIP extension voicemail, you should
first make sure the tested email to your email address works fine.
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Figure 3.4.5.1
3.4.6 Blacklist
Blacklist is used to block an incoming/outgoing call. If the number of
incoming/outgoing call is registered in the number blacklist, the caller will hear
the following prompt: “The number you have dialed is not in service. Please
check the number and try again”. The system will then disconnect the call.
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Figure 3.4.6
3.4.6.1 Create Blacklist
Click ‘New Blacklist’ to create a new number blacklist.
Figure 3.4.6.1
·Blacklist number
Enter the number you would like to block.
·Type
The number blocked for incoming or outgoing calls or both.
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3.5 Internal Settings
3.5.1 Options
1) General
·Ring Timeout
Number of seconds to ring a device before handling the call as per the
extension’s Follow Me settings . Default value is 30s.
.MAX call duration
The absolute maximum amount of time permitted for a call. A setting of 0
disables the timeout. Default value is 6000s.
.Maximum concurrent calls
Maximum concurrent calls limits. Default value 0 means no limit
·Music on hold
Used to set hold music for the system.
·Tone Region
Please select your country or nearest neighboring country to enable the default
dial tone, busy tone, and ring tone for your region.
Note: please reboot the system to take it effect.
·HTTP bind port/Web Access Port
Port to use for HTTP sessions . Default: 80
Note: please reboot the system to take it effect.
·FXO Mode
FXO port’s operation mode .
·Enable Follow Me Prompt
When set Follow me to Transfer to number on the extension page (e.g. when
500 is busy, transfer to 501), while 500 is busy, the call will be transferred to
501. If ‘Enable Follow Me Prompt’ choosing yes, there will be prompt before
transferring the call. Otherwise, the call will be transferred directly without any
prompt. Default: Yes.
·Virtual Ring Back Tone
It’s only for GSM trunk. Once enabled, when the caller call out with GSM trunks,
the caller will only hear the virtual ring back tone generated by the system
before callee answers the call.
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· Distinctive Caller ID
When incoming calls are routed from ring group/queue/IVR, the caller ID
displays with the name of ring group/queue/IVR ， for example
5503302(ringgroup_default)
Note: To display IVR's name, please press the key instead of the extension
number directly.
·Follow Me Prompt
Configure whether to play a prompt ‘please wait while trying to look at the
person you are calling’ when transfer a call by follow me settings.
·Invalid Phone Number Prompt
Configure the prompt when the dialed phone number is invalid.
·Busy Line Prompt
Configure the prompt when the dialed phone number is busy.
·Dial Failure Prompt
Configure the prompt when dial failed due to conjunction no-available channel.
2) Extension Preferences
·User Extensions
The default value is 500 to 616
·Ring Group Extensions
The default value is 620 to 629
·Paging Group Extensions
The default value is 630 to 639
·Conference Extensions
The default value is 640 to 659
·IVR Extensions
The default value is 660 to 679
·Queue Extensions
The default value is 680 to 689
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Figure 3.5.1
3.5.2 Business Hours
1) General
·Enable or Disable Business Hours
2) Others
·Enable Office Closed Timing
By dialing *81 (*81 is default) on an extension will force the office time closed
for the device whatever the general setting is.
·Disable Office closed timing
By dialing *081 (*081 is default) on an extension will disable the Office Closed
Timing.
3) Business Days
You can setup the business hours here.
4) Holidays
You can setup the holidays here.
If a time period is configured as both Holidays and office hours, it will be treated
as Holidays.
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Figure 3.5.2
3.5.3 Feature Codes
1) General
·One Touch Record
A user may initiate or stop call recording by dialing *1 during a call. (*1 is
default setting)
·Extension for Checking Voicemail
Users can check their Voicemail by dialing *2 on their phone (*2 is default
setting).
·Voicemail main menu
Users can go to the main menu by dialing *02 (*02 is default setting).
·Attended Transfer
Users may transfer an incoming call by dialing *3 on their phone (*3 is default
setting).
·Attended Transfer Timeout
The time out of transferring a call
·Blind Transfer
Users may blind transfer an incoming call by dialing *03 on their phone (*03 is
default setting).
·Call Pickup
Users may pick up an incoming call by dialing *4 on their phone (*4 is default
setting)
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·Extension Pickup
Users may pick up a specific extension’s incoming call by dialing *04+extension
number on their phone (*04 is default setting)
·Intercom
Define the feature code that is used to dial an extension in intercom mode. For
instance setting this value to *5 would allow you to initiate an intercom call with
extension 501 by dialing *5501.
·Normal Spy
In this mode, you can only listen to the extension being spied, for example you
can dial *90501 to monitor extension 501
·Whisper Spy
In this mode you can listen/whisper to the extension being spied, for example,
dialing *91501 to listen to extension 501, you can also talk with 501 too.
·Barge Spy
In this mode, you can barge in both extensions involved the call, for example
dialing *92501 to barge in and talk with all the extensions inside
2) Call Park Preferences
·Call Parking
User may park an incoming call on his own telephone by pressing ‘*6’ (*6 is
default setting)
·Extension range used to park calls
User may park an incoming call on a designated extension at first and then pick
up the call again on any other extension.
·Number of seconds a call can be parked before it is recalled.
Defines the number of seconds that a call can be parked before it is recalled to
the station that parked it .
3) Call Forwarding Preferences
·Reset to Defaults
Users may reset all call forward defaults by calling *70 on their phone (*70 is
default setting).
Note: When reset to defaults. The call forwarding settings will be configured as
follows:
Always forward: Disabled
Busy forward to Voicemail: Enabled
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No answer forward to Voicemail: Enabled
Do not disturb: Disabled
·Enable Forward All Calls
Users may enable always forward by calling *71 on their phone (*71 is default
setting)
·Disable Forward All Calls
Users may disable always forward by calling *071 on their phone (*071 is
default setting)
·Enable Forward When Busy
Users may enable busy forward by dialing *72 on their phone (*72 is default
setting)
·Disable Forward When Busy
Users may disable busy forward by calling *072 on their phone (*072 is default
setting)
·Enable Forward No Answer
Users may enable no answer forward by calling *73 on their phone (*73 is
default setting)
·Disable Forward No Answer
Users may disable no answer forward by calling *073 on their phone (*072 is
default setting)
·Forward to number
Users may activate call forwarding by dialing this feature code, followed by the
extension or phone number to forward all calls to.
Note: Users may activate Forward to number by dialing *74 + phone number.
e.g.: by dialing *74501, all calls will be forwarded to extension 501.
·Forward to Voicemail
Users may forward the call to Voicemail by calling *074 on their phone (*074 is
default setting)
·Enable Do Not Disturb
Users may enable do not disturb by calling *75 on their phone (*75 is default
setting)
·Disable Do Not Disturb
Users may disable do not disturb by calling *075 on their phone (*075 is default
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setting)
Figure 3.5.3
3.5.4 SIP Settings
1) General
·UDP Port
Port use for sip registrations, Default is 5060.
·TCP Port
Port use for sip registrations, Default is 5060.
·TLS Port
Port use for sip registrations, Default is 5061.
·RTP Port Start
Beginning of RTP port range
·RTP Port End
End of RTP port range
·DTMF Mode
Set default mode for sending DTMF. Default setting: rfc2833
·Max Registration/Subscription Time
Maximum duration (in seconds) of a SIP registration . Default is 3600 seconds.
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·Min Registration/Subscription Time
Minimum duration (in seconds) of a SIP registration . Default is 60 seconds.
·Default Incoming/Outgoing Registration Time
Default Incoming/Outgoing Registration Time: Default duration (in seconds)
of incoming/outgoing registration.
·Register Attempts
The number of SIP REGISTER messages to send to a SIP Registrar before giving
up. Default is 8 times.
·Register Timeout
Number of seconds to wait for a response from a SIP Registrar before timed out .
Default is 20 seconds.
·Calling Channel Codec Priority
Once enabled, when dialing out via SIP/SPS trunks, the codec of calling channel
will be selected in preference. If not, MyPBX will follow the priority in your
SIP/SPS trunks.
·Video Support
Support for SIP video or no. Default is yes.
·Max Bit Rate
Configure the max bit rate for video stream. The default: 384kb/s
·DNS SRV Look Up
Please enable this option when your SIP trunk contains more than one IP
address.
·User Agent
To change the useragent parameter of asterisk, the default is ‘MyPBX’, you
should change it if needed.
2) NAT
Note: Configuration of this section is only required when using remote
extensions.
·Enable STUN
STUN (Simple Traversal of UDP through NATs) is a protocol for assisting devices
behind a NAT firewall or router with their packet routing.
·STUN Address
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The STUN server allows clients to find out their public address, the type of NAT
they are behind and the internet side port associated by the NAT with a
particular local port. This information is used to set up UDP communication
between the client and the VOIP provider and so establish a call.
·External IP Address
The IP address that will be associated with outbound SIP messages if the system
is in a NAT environment .
·External Host
Alternatively you can specify an external host, and the system will perform DNS
queries periodically.
This setting is only required when your public IP address is not static. It is
recommended that a static public IP address be used with this system. Please
contact your ISP for more information.
·External Refresh Interval
If an external host has been supplied, you may specify how often the system will
perform a DNS query on this host. This value is specified in seconds.
·Local Network Identification
Used to identify the local network using a network number/subnet mask pair
when the system is behind a NAT or firewall .
Some examples of this are as follows:
'192.168.0.0/255.255.0.0' : All RFC 1918 addresses are local networks;
'10.0.0.0/255.0.0.0' : Also RFC1918;
'172.16.0.0/12':Another RFC1918 with CIDR notation;
'169.254.0.0/255.255.0.0' : Zero conf local network.
Please refer to RFC1918 for more information.
·NAT Mode
Global NAT configuration for the system . The options for this setting are as
follows:
Yes = Use NAT. Ignore address information in the SIP/SDP headers and reply to
the sender's IP address/port.
No = Use NAT mode only according to RFC3581.
Never = Never attempt NAT mode or RFC3581 support.
Route = Use NAT but do not include rport in headers.
·Allow RTP Reinvite
By default, the system will route media steams from SIP endpoints through
itself. Enabling this option causes the system to attempt to negotiate the
endpoints to route packets to each other directly, bypassing the system. It is not
always possible for the system to negotiate endpoint-to-endpoint media
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routing.
3）Codecs
A codec is a compression or decompression algorithm that used in the
transmission of voice packets over a network or the Internet.
u-law: A PSTN standard codec, used in North America, that provides very
good voice quality and consumes 64kbit/s in each direction (receiving and
transmitting) of a VoIP call.
a-law: A PSTN standard codec, used outside of North America, that
provides very good voice quality and consumes 64kbit/s in each direction
(receiving and transmitting) of a VoIP call.
GSM: A wireless standard codec, used worldwide, that provides adequate
voice quality and consumes 13.3kbit/s in each direction (receiving and
transmitting) of a VoIP call. GSM is supported by many VoIP phones.
SPEEX: Speex is an Open Source/Free Software patent-free audio
compression format designed for speech. The Speex Project aims to lower
the barrier of entry for voice applications by providing a free alternative to
expensive proprietary speech codecs. Moreover, Speex is well-adapted to
Internet applications and provides useful features that are not present in
most other codecs.
G.722: G.722 is a wideband speech coding algorithms which supports the
bit rate of 64, 56 and 48kbps wideband. It’s a broadband voice encoding of
G series.
G.726: A PSTN codec, used worldwide, that provides good voice quality and
consumes 32kbit/s in each direction (receiving and transmitting) of a VoIP
call. G.726 is supported by some VoIP phones.
ADPCM, G.729A, H261, H263, H263p, H264,MPEG4.
Note: If you would like to use G.729, please enter your license.
4）QOS
QoS (Quality of Service) is a major issue in VOIP implementations. The issue
is how to guarantee that packet traffic for a voice or other media connection
will not be delayed or dropped due interference from other lower priority
traffic.When the network capacity is insufficient, QoS could provide priority
to users by setting the value.
5) Advanced Settings
·From Field
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Where to get the caller ID in sip packet.
·To Field
Where to get the DID in sip packet.
·180 Ringing
It is set when the telecom provider needs. Usually it is not needed.
·Qualify
Send check alive packets to the sip provider.
·Remote Party ID
Whether send Remote-Party-ID on SIP header. Default no.
·Session -timers
Enable sesstion-timer mode, default: yes
·Sesstion-expires
The max refresh interval
·Sesstion-minse
The min refresh interval, which mustn't be less than 90s
·Sesstion-refresher
Choose sesstion-refersher, the default is Uas
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Figure 3.5.4
3.5.5 IAX Settings
1) General
·Bind Port
Port use for IAX2 registrations, Default is 4569.
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·Bandwidth
Low/medium/high with this option you can control which codec to be used.
·Min Registration Time
Minimum duration (in seconds) of a IAX2 registration. Default is 60 seconds.
·Max Registration Time
Maximum duration (in seconds) of a IAX2 registration. Default is 1200 seconds.
2）Codecs
A codec is a compression or decompression algorithm that used in the
transmission of voice packets over a network or the Internet.
u-law: A PSTN standard codec, used in North America, that provides very
good voice quality and consumes 64kbit/s in each direction (receiving and
transmitting) of a VoIP call.
a-law: A PSTN standard codec, used outside of North America, that
provides very good voice quality and consumes 64kbit/s in each direction
(receiving and transmitting) of a VoIP call.
GSM: A wireless standard codec, used worldwide, that provides adequate
voice quality and consumes 13.3kbit/s in each direction (receiving and
transmitting) of a VoIP call. GSM is supported by many VoIP phones.
SPEEX: Speex is an Open Source/Free Software patent-free audio
compression format designed for speech. The Speex Project aims to lower
the barrier of entry for voice applications by providing a free alternative to
expensive proprietary speech codecs. Moreover, Speex is well-adapted to
Internet applications and provides useful features that are not present in
most other codecs.
G.726: A PSTN codec, used worldwide, that provides good voice quality and
consumes 32kbit/s in each direction (receiving and transmitting) of a VoIP
call. G.726 is supported by some VoIP phones.
ADPCM, G.729A, H261, H263, H263p, H264.
Note: If you would like to use G.729, please enter your license.
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Figure 3.5.5
3.5.6 Voicemail Settings
1) General Voicemail Settings
a) Message Options
·Max Messages per Folder
Set the maximum number of messages that can be stored in a single voicemail
box.
·Max Message Time
Set the maximum length of a single voicemail message.
·Min Message Time
Set the minimum length of a single voicemail message. Messages below this
threshold will be automatically deleted.
·Ask Caller to Dial 5
If this option is set, the caller will be prompted to press 5 before leaving a
message.
·Operator Breakout from Voicemail
If this option is set, the caller can jump out of the voicemail and go to the
destination (IVR) you set by dialing “0”.
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b) Greeting Settings
·Busy Prompt
Greeting played when the extension called is busy.
Skip greeting: Do not play a greeting.
Play busy greeting: play the extension busy greeting.
·Unavailable Prompt
Greeting played when the extension called is Unavailable.
Skip greeting: Do not play a greeting.
Play Unavailable greeting: play the extension Unavailable greeting.
·Leave a Message Prompt
Greeting when ask the caller to dial 5 to leave a message.
Skip greeting: Do not play a greeting.
Play busy greeting: play the extension busy greeting.
Play Unavailable greeting: play the extension Unavailable greeting.
c) Playback Options
·Announce Message Caller ID
If this option is enabled, the Caller ID of the party that left the message will be
played back before the voicemail message begins playing.
·Announce Message Duration
If this option is set, the duration of the message in minutes will be played back
before the voicemail message begins playing.
. Announce Message Arrival Time
If this option is set, the arrival time of the message will be played back before
the voicemail message begins playing.
. Allow Users to Review Messages
Allow callers to review their recorded message before sending it to voicemail.
2) SMTP Settings for Voicemail
Note: If you want to send voicemail messages as email attachments, please
configure this section.
·E-mail Address
The E-mail Address that MyPBX will use to send voicemail.
·Password
The password for the email address used above
·SMTP Server
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The IP address or hostname of an SMTP server that the MyPBX will connect to in
order to send voicemail messages via email, i.e. mail.yourcompany.com.
·Port
SMTP Port: the default value is 25.
·Use SSL/TLS to send secure message to server
If the server of sending email needs to authenticate the sender, you need to
select the check box.
Note: Must be selected for Gmail or exchange server.
After filling out the above information, you can click on the ‘Test Account
Settings’ button to check whether the setup is OK.
1) If the test is successful, you can use the email safely.
2) If test failed, please check the above information is correct or network is
proper.
Figure 3.5.6
3.5.7 SMS Settings
MyPBX supports SMS to Email and Email to SMS. (Only for GSM trunks)
1) Enable SMS to Email
If you enable this, as soon as the GSM trunks receive small messages, MyPBX
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will send the text of this message to the email addresses listed on the Email List.
You can add email addresses to the Email List.
Figure 3.5.7.1
2) Enable Email to SMS
If you enable this, you can use MyPBX to send out message by sending an email
to the specified address.
·Enable Country Code
If you want to add country code before the dialed numbers, please tick this.
·Country Code
The country code to be add before the dialed numbers.
·Receive mails every
The intervening time of receiving mails from POP3 server.
·Access Code
This PIN code is used to verify the subject of the emails received. If the form of
email passes the verification, it will be send out by SIM card. If not, this email
will be deleted immediately.
3) Email Settings
Note:
1. If you want to use “SMS to Email”, please configure POP3 setting.
2. If you configure the POP3 setting, MyPBX will download emails from the mail
server regularly. Once downloaded, the emails will be deleted from the mail
server.
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·Email Address
This email address will be used to:
1. Send email to the addresses listed on “SMS to Email” setting.
2. Receive email and send the text of the email to the target mobile number by
SMS.
Note: If you use gmail, just put your user name here. E.g. email address:
test@gmail.com, you just put “test” here.
·Password
Input the password of this email here.
·SMTP Server (SMTP)
·SMTP Server Port
·Receive Server (POP3)
·Receive Server Port
If you want to know more about Email to SMS, please refer to APPENDIX F
Figure 3.5.7.2
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3.5.8 DISA
DISA (Direct Inward System Access) allows someone calling in from outside the
telephone switch (PBX) to obtain an ‘internal’ system dial tone and make calls as
if they were using one of the extensions attached to the telephone switch. To use
DISA, a user calls a DISA number, which invokes the DISA application. The DISA
application in turn requires the user to enter a PIN number, followed by the
pound sign (#). If the PIN number is correct, the user will hear dial tone on
which a call may be placed. Obviously, this type of access has serious security
implications, and great care must be taken not to compromise your security.
Figure 3.5.8.1
To add a new DISA application, click the New DISA button.
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Figure 3.5.8.2
1) General
·DISA Name
Give this DISA application a name to help you identify it.
·PIN #
The password for this DISA .
·Response Timeout
The maximum amount of time the system will wait before hanging up the call if
the user has dialed an incomplete or invalid number. Default is 10 seconds.
·Digit Timeout
The maximum amount of time permitted between each digit when the user is
dialing an extension number. Default is 5 seconds.
2) Member Outbound Routes
Used to set the outbound routes that can be accessed from this DISA .
3.5.9 Conferencing
Conference Calls increase employee efficiency and productivity, and provide a
more cost-effective way to hold meetings. Conference agents can dial * to
access to the settings options and the admin can kick the last user out and can
lock the conference room.
·Extension
This is the number dialed to reach this Conference Room.
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·Admin
Admin can kick a user out and can lock the conference room.
·Pin #
Set a PIN # that must be entered in order to access this conference room (i.e.
1234).
Figure 3.5.9
3.5.10 Paging Groups
Paging is used to make an announcement over the speakerphone to a phone or
group of phones. Targeted phones will not ring, but instead answer immediately
into speakerphone mode. Please note that this section is for configuring paging
groups. If you would like to configure Intercom settings, please open the Other
Settings -> Feature Codes screen.
This feature is supported by the following SIP phones:
Yealink's T28, T26, T22, T20, T10T, T9CM. Other SIP devices may also work
with this feature but are not officially supported.
Note: A paging group can have a maximum of 20 members.
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Figure 3.5.10.1
·Paging Group Number
Defines the numbered extension that may be dialed to reach this group.
·Duplex
Paging is typically one way for announcements only. Checking this will make
paging duplex, allowing all users in the group to talk and be heard by all.
Figure 3.5.10.2
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3.5.11 DNIS Settings
DNIS (Dialed Number Identification Service) is a telephone service that
identifies for the receiver of a call the number that the caller dialed.
Note:
1. DID number is not available in PSTN/GSM/UMTS trunks
2. If DID is not configured here, all the calls via this trunk will show the DNIS
instead of the original caller ID
Figure 3.5.11
3.5.12 PIN User Settings
PIN User are used to manage lists of PINs that can be used to access restricted
features such as Outbound Routes. If user use PIN User call out, system will
auto record the PIN in the call detail records.
1) Options
·Access Code
.Dial this code to access PIN.
·Prompt for Entry
Prompt caller enter the PIN Number.
·Prompt for Entry Failure
Prompt the caller when an invalid PIN is entered.
2) PIN User
MyPBX can store a number of PIN Users. PIN Users may be used to keep track
of calls in relation to particular activities or clients. They can also be used to
keep track of calls by particular users or sets of users.
• PIN entered are checked against those stored by the system. If an invalid PIN
is entered, the PIN is requested again.
• The system administrator can configure certain numbers or types of numbers
to require entry of a PIN before you can continue making a call to such a
number.
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• The system administrator can also configure you to have to enter a PIN before
making any external call.
·Name
A character-based name for this PIN list, i.e. ‘YeastarPIN’
·PIN List
Enter a list of one or more PINs, One PIN per line.
·Outbound Route
PIN User can use those outbound route to make call out.
Figure 3.5.12
3.5.13 Callback Settings
MyPBX allows caller A to dial an inbound route number, and after hearing the
ring, A can hang up the call or wait for MyPBX to cut off the call, then MyPBX will
call A with this number. When A pick up the call, A can dial the number he wants
to call; MyPBX will call the number with its outbound route.
Note:
1. If you’d like to use callback feature, please make sure if it’s enabled on the
inbound route setting panel.
2. No callback rules needed to be set if the trunk supports call back with the
caller ID directly.
·Allow All Numbers
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If you want to apply Callback function to all incoming numbers, please tick Allow
All numbers.
Follow the step to use this function.
Step 1: Enable Callback.
Inbound Routes – Choose “Yes” on” Enable Callback” to enable this function.
Refer to chapter 3.4.5.1
Figure 3.5.13.1
Step 2: Create Callback number
Figure 3.5.13.2
Step 3: Create Callback Rules
You will need to create callback rules when the system should strip or add digits.
·Trunk Name
Choose the trunk with callback rules
·Strip digits from front
Define how many digits will be stripped from the call in number before the
callback is placed. For example, when you call from number 123456789 into
MyPBX, the caller ID is 0123456789, but you can only call 123456789
successfully from MyPBX trunk. You should configure number 0123456789 as
the call back number and strip 1 digit before the callback is placed
·Prepend before dialing
Define digits added before a callback number before the callback is placed. For
example, the call in number (Caller ID) is 123456789, MyPBX need to send
9123456789 to its trunk when call to this number. You should configure
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123456789 as the call back number and add 9 before the callback is placed. You
can add ‘w’ for analog trunks for some delay too.
Figure 3.5.13.3
3.5.14 Speed Dial Settings
1) Options
·The prefix of speed dial
The prefix should be dialed before the speed dial number. Default is *99.
2) Add new speed dial.
·Source Number
The speed dial number.
·Destination Number
The number you want to call.
e.g. The source number is ”123”. The destination number is 5503305.The prefix
number is *99.You can use an extension with any type to dial *99123, then it
will call to number 5503305.
Note: Don’t forget to add the outbound dial prefix if you would like to dial the
speed dial number through trunk.
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Figure 3.5.14
3.5.15 Music on Hold Prompts
The administrator can upload on hold music as follows:
1) Click ‘Upload Music on Prompt ’
2) Click ‘Browse’ to choose the desired audio file.
3) Click ‘Upload’ to upload the selected file.
Note: The sound file format should be as follows: GSM 6.10, 8.000kHz, Mono,
1kb/sec, The file size must not be greater than 1.8MB.
Figure 3.5.15
3.6 Network Settings
3.6.1 LAN Settings
·DHCP
If this option is set, MyPBX will use DHCP to get an available IP address from
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your local network. Not recommended.
·Enable SSH
This is the advance way to access the device, you can use the putty software
to access the device. In the SSH access, you can do more advance setting
and debug.
·Port: the default is 8022,
·Hostname
Set the host name for MyPBX.
·IP Address
Set the IP Address for MyPBX.
·Subnet Mask
Set the subnet mask for MyPBX.
·Gateway
Set the gateway for MyPBX.
·Primary DNS
Set the primary DNS for MyPBX.
·Secondary DNS
Set the secondary DNS for MyPBX.
·IP Address2
Set the second IP Address for MyPBX.
·Subnet Mask2
Set the second subnet mask for MyPBX.
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Figure 3.6.1
3.6.2 WAN Settings
Note: Only have WAN port’s hardware version support this functions.
It support three connection types: DHCP (obtain an IP automatically), PPPoE,
Static IP Address.
·DHCP
.If your ISP says that you are connecting through DHCP or a dynamic IP address
from your ISP, perform these steps:
Step1: Select DHCP as the WAN Connection Type.
Step2: Click Save button to save the settings.
Step3: Reboot the device.
Step4: Check the WAN’s Status (System Info  WAN  Status).
·Static IP Address
If your ISP says that you are connecting through a static or fixed IP address
from your ISP, perform these steps:
Step1: Select Static IP Address as the WAN Connection Type.
Step2: Enter the IP Address.
Step3: Enter the Subnet Mask.
Step4: Enter the Gateway Address.
Step5: Enter the Primary DNS and Secondary DNS.
Step6: Click the Save button to save the settings.
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Step7: Reboot the device.
Step8: Check the WAN’s Status (System Info  WAN  Status).
·PPPoE
If your DSL provider says that you are connecting through PPPoE or if you
normally enter a user name and password to access the Internet, perform these
steps:
Step1: Select PPPoE as the WAN Connection Type.
Step2: Enter the User Name.
Step3: Enter the Password.
Step4: Click the Save button to save the settings.
Step5: Reboot the device.
Step6: Check the WAN’s Status (System Info  WAN  Status).
Figure 3.6.2
3.6.3 Static Route
MyPBX will have more than one internet connection in some situations but it has
only one default gateway. You will need to set some Static Route for MyPBX to
force it goes out through different gateway when access to different internet.
The default gateway priority of MyPBX from high to low is OpenVPNWAN
portLAN port.
1) Route table
The current route rules of MyPBX
·Destination
The destination network to be accessed to by MyPBX
·Subnet Mask
Specify the destination network portion.
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·Gateway
Which gateway MyPBX will go through when access to the destination network.
·Metric
The cost of a route is calculated by using what are called routing metric. Routing
metrics are assigned to routes by routing protocols to provide measurable
statistic which can be used to judge how useful (how low cost) a route is.
·Interface
Define which internet port to go through.
2) Static Route Rules
You can add new static route rules here.
Figure 3.6.3
3.6.4 Firewall
Firewalls are used to prevent unauthorized Internet users from accessing
private networks connected to the Internet, especially intranets. All
messages entering or leaving the intranet pass through the firewall, which
examines each message and blocks those that do not meet the specified
security criteria.
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Figure 3.6.4
1) Enable Firewall
Enable the firewall to protect the device. You should reboot the device to let
the firewall run successfully.
2) Common Rules
·Name
A name for this rule , e.g. ‘HTTP’.
·Description
Simple description for this rule . E.g.: Accept the specific host to access the web
interface for configuration.
·Protocol
The protocols for this rule .
·Port
Initial port should be on the left and end port should be on the right.
The end port must be equal to or greater than start port.
·IP
The IP address for this rule . The format of IP address is: IP/mask
Ex: 192.168.5.100/255.255.255.255 for IP 192.168.5.100
Ex: 216.207.245.47/255.255.255.255 for IP 216.207.245.47
Ex:192.168.5.0/255.255.255.0 for IP from 192.168.5.0 to 192.168.5.255 .
·MAC Address
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The format of MAC Address is XX:XX:XX:XX:XX:XX, X means 0~9 or A~F in hex,
the A~F are not case sensitive.
·Action
Accept: Accept the access from remote hosts.
Drop: Drop the access from remote hosts.
Ignore: Ignore the access.
Figure 3.6.4.1
3) Auto Defense
·Port
Auto defense port, e.g.: 8022.
·Protocol
Auto defense protocol , TCP or UDP.
·Rate
The maximum packets or connections can be handled per unit time.
E.g.: (Port: 8022 Protocol: TCP Rate: 10/minute) means maximum 10 TCP
connection to port 8022 can be handled per minute, the eleventh connection will
be refused directly.
Figure 3.6.4.2
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4) IP Blacklist
You can set some packets accept speed rules here. When a IP address which
hasn’t been accepted in common rules sends packets faster than the allowed
speed, it will be set as black IP address and blocked automatically.
a) New Rule
·Port
Auto defense port
·Protocol
Auto defense protocol. TCP or UDP.
·IP Packets
Allowed IP packets number in the specific time interval.
·Time interval
The time interval to receive IP packets. For example, IP packets 90,time
interval 60 means 90 IP packets are allowed in 60 seconds.
Figure 3.6.4.3
b) IP Blacklist Manage
You can manage the IP addresses which are blocked automatically here.
Figure 3.6.4.4
5) Other Settings
·Disable Ping
Enable this item, net ping from remote hosts will be dropped.
·Drop All
When you enable ‘Drop All’ feature, system will drop all packets or connection
from other hosts if there are no other rules defined. To avoid locking the devices,
at least one ‘TCP’ accept common rule must be created for port used for SSH
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access, port used for HTTP access and port sued for CGI access.
3.6.5 DHCP Server
Dynamic Host Configuration Protocol (DHCP) is a network protocol that enables
a server to automatically assign an IP address to a computer from a defined
range of numbers (i.e., a scope) configured for a given network. You can set a
local network NTP server for MyPBX here too.
Figure 3.6.5
3.6.6 VLAN Settings
A VLAN(Virtual LAN) is a logical local area network (or LAN) that extends beyond
a single traditional LAN to a group of LAN segments, given specific
configurations.
1) VLAN Over Lan
·NO.1
Click the NO.1 you can edit the first VLAN over Lan.
·VLAN Number
.The VLAN Number is a unique value you assign to each VLAN on a single device.
·VLAN IP Address
Set the IP Address for MyPBX VLAN over Lan.
·VLAN Subnet Mask
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Set the Subnet Mask for MyPBX VLAN over Lan.
·Default Gateway
Set the Default Gateway for MyPBX VLAN over Lan
·NO.2
Click the NO.2 you can edit the first VLAN over Lan.
·VLAN Number
.The VLAN Number is a unique value you assign to each VLAN on a single device.
·VLAN IP Address
Set the IP Address for MyPBX VLAN over Lan.
·VLAN Subnet Mask
Set the Subnet Mask for MyPBX VLAN over Lan.
·Default Gateway
Set the Default Gateway for MyPBX VLAN over Lan.
2) VLAN Over Wan
·NO.1
Click the NO.1 you can edit the first VLAN over Wan.
·VLAN Number
.The VLAN Number is a unique value you assign to each VLAN on a single device.
·VLAN IP Address
Set the IP Address for MyPBX VLAN over Wan.
·VLAN Subnet Mask
Set the Subnet Mask for MyPBX VLAN over Wan.
·Default Gateway
Set the Default Gateway for MyPBX VLAN over Wan.
·NO.2
Click the NO.2 you can edit the first VLAN over Wan.
·VLAN Number
.The VLAN Number is a unique value you assign to each VLAN on a single device.
·VLAN IP Address
Set the IP Address for MyPBX VLAN over Wan.
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·VLAN Subnet Mask
Set the Subnet Mask for MyPBX VLAN over Wan.
·Default Gateway
Set the Default Gateway for MyPBX VLAN over Wan.
Figure 3.6.6
3.6.7 VPN Settings
A virtual private network (VPN) is a method of computer networking--typically
using the public internet--that allows users to privately share information
between remote locations, or between a remote location and a business' home
network. A VPN can provide secure information transport by authenticating
users, and encrypting data to prevent unauthorized persons from reading the
information transmitted. The VPN can be used to send any kind of network
traffic securely. MyPBX supports OpenVPN.
·Enable VPN
·Import VPN Config
Import configuration file of OpenVPN. Don't configure 'user' and 'group' in the
'config' file.
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Figure 3.6.7
3.6.8 DDNS Settings
DDNS(Dynamic DNS) is a method / protocol / network service that provides the
capability for a networked device, such as a router or computer system using
the Internet Protocol Suite, to notify a Domain Name System (DNS) name
server to change, in real time, the active DNS configuration of its configured
hostnames, addresses or other information.
·Enable DDNS
·DDNS Server
Select the DDNS server you sign up for service.
·User Name
User name the DDNS server provides you.
·Password
User account’s password .
·Host Name
Note: DDNS allows you to access your network using domain names instead of
IP address. The service manages changing IP address and updates your domain
information dynamically. You must sign up for service through dyndns.org,
freedns.afraid.org, www.no-ip.com, www.zoneedit.com
Figure 3.6.8
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3.7 System Settings
3.7.1 External Storage
The External Storage feature is used to extend storage space. Once configured,
the files (voicemail, call recording files, CDR files) created before the configured
days will be moved to the Net-Disk.
Auto-Backup extends the allocated disk space for backing up critical files. When
properly configured, MyPBX will move all qualified files to a Windows PC every
30 minutes. For the voicemail files and recoding files, they must be created
before Auto-Backup has been configured.
How to configure External Storage, please see APPENDIX B How to Configure
Autobackup.
Figure 3.7.1
3.7.2 Password Settings
The default password is ‘password’. To change the password, enter the new
password and click update. The system will then prompt you re-login using your
new password.
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Figure 3.7.2
3.7.3 System Prompts
MyPBX have prompts of many languages. You can download the appropriate
language you need. MyPBX can support American English, Australian English,
Chinese, Dutch, French, Canadian French, German, Greek, Hungarian, Italian,
Polish, Portuguese, Brazilian Portuguese, Russian, Spanish, Mexican Spanish,
Turkish, Thai, Korean currently.
Note:
Auto-detection is highly recommended. But if you prefer to download via HTTP
or TFTP server, please contact the local dealer for the prompts.
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Figure 3.7.3
3.7.4 Date and Time
Set the date and time for MyPBX.
·Time Zone
You can choose your time zone here.
·Daylight Saving Time
Set the mode to Automatic or disabled
·Automatically Synchronize With an Internet Time Server
Input the NTP server so that MyPBX will update the time automatically
·Set Date & Time Manually
Figure 3.7.4
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3.7.5 Backup and Restore
Figure 3.7.5
3.7.6 Reset and Reboot
·Reboot System
Warning: Rebooting the system will terminate all active calls!
·Reset to Factory Defaults
Warning: A factory reset will erase all configuration data on the system.
Please do not turn off the system until the RUN light begins blinking. Any power
interruption during this time could cause damage to the system.
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Figure 3.7.6
3.7.7 Firmware Update
Upgrading of the firmware is possible through the Administrator web interface
using a TFTP Server or an HTTP URL.
Enter your TFTP Server IP address and firmware file location, then click start to
update the firmware.
More Information, please see http://www.yeastar.com/download/MyPBX/
Note：
1. If enabled ‘Reset configuration to Factory Defaults’, System will restore to
factory default settings.
2. When update the firmware, please don’t turn off the power.
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Figure 3.7.7
3.8 Reports
3.8.1 Call Logs
The call Log captures all call details, including call time, caller number, callee
number, call type, call duration, etc. An administrator can search and filter call
data by filter the call logs by call date, caller/callee, trunk, duration, billing
duration, status, communication type.
Figure 3.8.1
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3.8.2 System Logs
You can download and delete the system logs of MyPBX.
Options
·Enable Hardware Log
Save the infomation of hardware; (up to 4 log files)
·Enable Normal Log
Save the prompt information; (up to 16 log files)
·Enable Web Log
Save the history of web operations (up to 2 log files)
·Enable Debug Log
Save debug information (up to 2 log files)
Figure 3.8.2
3.8.3 System Info
General:
Information about hardware version , firmware version and system uptime.
LAN:
Information about hostname, MAC address, IP address, gateway, Primary
DNS and Secondary DNS.
Disk Usage:
Disk usage information.
Memory Usage:
Memory usage information .
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Figure 3.8.3
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4. Access MRI
MRI (MyPBX Recording Interface).Users may access MRI by logging into the MRI
web interface with their username (extension number) and voicemail password.
4.1 Allow users to access MRI
The extension’s ‘User Web Interface’ option must be checked before the
associated user can log into MRI.
Figure 4-1
4.2 User login
Users can access the MRI web interface by navigating to the MyPBX IP address
using a web browser. If you are unsure of this address, please contact your
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network administrator.
Figure 4-2
4.3 Voicemail
Users can check, delete, move and download voicemail files here.
Figure 4-3
4.4 Record
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Users can play, delete and download recorded files here.
Figure 4-4
4.5 Voicemail Settings
·Voicemail password: new voicemail box password.
·Enter again to confirm: confirm new voicemail box password.
·Email Address: Email address use to receive the voicemail or Fax.
Note: Please ensure that the section 'SMTP Settings For Voicemail'(in the
'Voicemail Settings') have been properly configured before using this feature.
·Enable Voicemail
Check this box if the user should have a voicemail account.
·Enable Send Voicemail
Once enabled, the voicemail or Faxex will be sent to email as an attachment.
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Figure 4-5
4.6 Settings
You can do some basic setting here. Such as call forwarding, DND, Mobile
Extension Number.
Figure 4-6
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5. Use MyPBX
GSM
PSTN
PSTN
IP
FXS
LAN
Route
601
602
603
604
IP
Soft
Normal
5.1 Make outbound call
To make an outbound call, we need to add trunk first. There are five types of
VoIP Trunk:
·Analog Trunk: FXO ports of MyPBX, connected to a local PSTN.
·GSM Trunk: GSM ports of MyPBX, connected to GSM Network.
·BRI Trunk: BRI ports of MyPBX, connected to a local PSTN.
·VoIP Trunk: Connected to remote VOIP service server.
·Service Provider: Connected to service provider server.
What are FXO and FXS?
FXS (Foreign exchange Station) is an interface which drives an analog
telephone or FAX machine. FXS interfaces deliver power, provide ringing, and
use FXO signaling. FXS interfaces are what allow you to hook telephones and
other analog devices to your PBX
FXO (Foreign exchange Office) is an interface that connects to a phone line to
supply your PBX with access to a public telephone network. FXO interfaces use
FXS signaling. FXO interfaces allow you to connect your PBX to real analog
phone lines.
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5.1.1 Sample Routing via PSTN Trunk
Let’s route all inside extensions through an analog trunk by dialing 9. In
Outbound Routes, add a new outbound route as below.
Figure 5-1
As we can see from the outbound route of ‘pstnout’, all phone numbers
starting with 9 will have their first digit stripped off (digit 9) and will be sent
to the PSTN (port 1-2).
After we have configured the above, we can dial 9 + local number to dial out
via a PSTN line.
Note: Setting number prefix to wild card X and setting Strip to 0 digits from
the front will allow all calls to go through this outbound route.
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5.1.2 Sample Routing via VoIP Trunk
Let’s configure all inside extensions to dial ‘0’ through the VoIP Trunk.
1. Add VoIP service provider
Before we do add this, please make sure you have a VoIP Trunk account.
Trunks VoIP Trunk  SIP Trunk
Enter your account information on this page, and click Save.
Figure 5-2
2. Add Outbound Routes
As we can see from the Outbound Route of ‘voipout’, all phone numbers
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starting with 0 will have their first digit stripped off (digit 0) and will be sent
to the SIP Trunk.
Figure 5-3
Now that we have added two outbound dialing rules, any call starting with 9
will be routed to the PSTN, and any number starting with 0 will be routed to
the SIP Trunk.
5.2 Incoming call
5.2.1 Sample Routing to an IVR
Let’s configure an incoming call to route to the IVR. In the IVR itself, let’s
configure digit 0 to route the call to extension 500, and digit 1 to route the
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call to extension 501.
1. Add IVR
To add a new IVR, go to IVR Create New IVR
Figure 5-4
2. Add Inbound Routes
As we can see from the Inbound Route of ‘allin’, all incoming calls will be sent
to the IVR.
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Figure 5-5
APPENDIX A FAQ
Q1. How to Register SIP device？
A1:
1) Register SIP soft phone
Download the x-lite softphone from counterpath website
www.counterpath.com
After install the x-lite, right click the panel and select the SIP Account setting
and then configure it.
Display Name: 500
User Name: 500
Password: 500
Authorization Name: 500
Domain: 192.168.5.150
2) Register IP Phone (for example, Yealink’s T28 IP Phone)
a) Connect the T28’s Internet port to the switch. And it can get the IP from
your route.
b) Press the ‘OK’ key on T28 to get the IP of T28.
c) Put the IP on web browser then you can enter the T28 configure page
through this IP.
d) Put the SIP extensions info on the T28 IP phones.
Display Name: 501
User Name: 501
Register Name: 501
Password: 501
SIP Server: 192.168.5.150
Use the same method register another T28 to other extension.
Q2. How do I reset MyPBX back to the factory default settings?
A2: To perform a reset, please follow steps below:
Step 1: Hold down the ‘Reset’ button on the back of the unit for 5 seconds
and watch the LEDs on the front of the MyPBX. When the status LED turns
red, let go of the reset button.
Step 2: When the RUN status LED starts blinking, MyPBX will be set back to
factory defaults.
Step 3: To access the configuration page, navigate to 192.168.5.150 using
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a web browser. Make sure that you are on the 192.168.5.0 subnet before
doing this.
Step 4: Login to the device with the username ‘admin’ and the password
‘password’, in order to begin reconfiguring the device.
APPENDIX B How to Configure
Autobackup
Before Autobackup can be properly configured, an SMB share folder
accessible from MyPBX must be set up on a Windows based machine. Once
that has been set up, please follow the steps below.
Step 1 Add a new folder, rename it, and set this new folder’s share
Properties according to Figure B-1
Figure B-1 Set up share Properties
Step 2 Enter the new folder and create a new text file, then rename this file
to status.txt. This step is very important, DO NOT forget to create the
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status.txt file.
Step 3 Configure Autobackup settings on MyPBX to Figure B-2
Figure B-2 Autobackup Setting
Net-Disk Host/IP: Change this to the IP address of the computer where
backup files will be stored.
Net-Disk Share Name: Change this to the name of the shared folder
where backups will be stored.
Net-Disk Share Username: The user name used to log into the network
share. Leave this blank if it is not required
Net-Disk Share Password: The password used to log into the network
share. Leave this blank if it is not required
If configuring is correctly, open your Windows share folder to see if the
MyPBX backup files and folders has been created. If the contents of the
backup folder look similar to Figure B-3, then you have successfully
configured Autobackup on the MyPBX unit.
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Figure B-3 Autobackup setting succeed
APPENDIX C How to Configure
NAT setting
When MyPBX is behind a NAT(firewall),you need to configure NAT
setting on MyPBX if you want to use a remote extension.
Please follow section 1 or 2 below depending on your network configuration.
1. If MyPBX is connected to a local network, you must set up port forwarding
on your router. Specifically, you must map port 5060 (default SIP port) and
port 10001-10200 (default RTP port range) as UDP ports.
Next, go to the MyPBX web interface and configure the SIP settings
according to Figure C-1:
External IP Address: your router’s public IP address
External Host: your router’s domain
External Refresh Interval: 20 seconds
Local Network Identification:192.168.5.0/255.255.255.0 (change this
according to your network configuration)
NAT mode: Yes
Allow RTP Reinvite: No
Figure C-1
Assuming that your router’s host address is yeastar.3322.org, your local
network is from 192.168.5.1-192.168.5.254, and the subnet Mask is
255.255.255.0, the MyPBX network settings should configured like Figure
C-2
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Figure C-2 MyPBX Network setting
2. If MyPBX has a public IP, (i.e. is connected directly to your internet service
provider), the network settings should be configured according to Figure
C-3:
Figure C-3
Next, you should configure the NAT settings according to Figure C-4
Figure C-4
External IP Address: The public IP address of MyPBX
External Host: Leave this blank if no domain has been configured
External Refresh Interval: Leave this blank
Local Network Identification: Leave this blank
NAT mode: Yes
Allow RTP Reinvite: No
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APPENDIX D How to Use Auto
Provision
Step1. Disable DHCP Server on your local network.
E.g. Disable DHCP Server on Linksys Router.
Figure D-1
Step2. Enable DHCP Server on MyPBX.
Login MyPBX web interface, System Settings  DHCP Server  Enable DHCP
Server.
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Figure D-2
Step3. Configure phones on MyPBX auto-provision page.
1. Login MyPBX web interface, Basic  Auto Provision  Create New Phone.
Figure D-3
2. Fill in the phone detail message on the pop-up windows.
Input IP Phone’s MAC address, configure Name, Call waiting, Line, Extension,
Label, Line active for the phone.
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Figure D-4
Step4. Turn on the power and connect the network cable to IP Phone.
Remark: The factory default setting of DHCP for IP Phone is enable, so you can
skip this step to step 5.
If the DHCP is disable, please follow below step to enable it. (e.g.: Yealink’s IP
Phone).
1. Login IP phone’s web page.
2. Enable DHCP.
Figure D-5
Step5. Finish.
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APPENDIX E How Do I Configure
Distinctive Ring Tones
Step1: On your IP phone, navigate to the Phone settings web configuration
page and find the Distinctive Ring Tone section.
For each custom ring tone, enter the Internal Ringer Text (can be digits or text)
to trigger the ring tone. For example, you may enter “Family”.
e.g.: Yealink’s IP phone.
Figure E-1
Step2. Configure the ‘Distinctive Ringtone’ on MyPBX.
MyPBX web interface, Inbound Routes  Edit Inbound Route, fill in the Internal
Ringer Text on ‘Distinctive Ringtone’.
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Figure E-2
Step3. Finish.
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APPENDIX F How to Use Email to
SMS
How to use Email to SMS
You need to send an email to the specified email address (you set in Email
Settings. In this case, it is lears@yeastar.com).The content of this email will be
sent to the number you want as message. The subject (title) of the email will
determine the number. Here are some examples of the formats to the subject of
the email.
Example:
1. Send message with no PIN code and default GSM port.
Format: phonenumber
if the subject is "12345678", the text of this email(“Welcome to Yeastar!”) will
be sent to number "12345678" through the first available GSM trunk(No pin
code should be set by administrator).
Figure F-1
2. Send message with no PIN code and specified GSM port.
Format: port:portnumber-phonenumber
if the subject is "port:9-12345678", the text of this email (“Welcome to
Yeastar!”) will be sent to the number "12345678" through GSM trunk 9 (No pin
code should be set by administrator).
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Figure F-2
3. Send message with PIN code and default GSM port.
Format: 500:pincodenumber-phonenumber
if the subject is "500:987-12345678", the text of this email(“Welcome to
Yeastar!”) will be sent to number "12345678" through the first available GSM
trunk("987" is the pin code set by administrator).
Figure F-3
4. Send message with PIN code and specified GSM port.
Format: 500:pincodenumber-port:portnumber-phonenumber
if the subject is "500:987-port:9-12345678", the text of this email(“Welcome to
Yeastar!”) will be sent to number "12345678" through GSM trunk 9("987" is
the pin code set by administrator).
Figure F-4
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APPENDIX G How to Use DID
Direct inward dialing (DID), also called direct dial-in (DDI) in Europe and
Oceania, is a feature offered by telephone companies for use with their
customers' private branch exchange (PBX) systems. In DID service the
telephone company provides one or more trunk lines to the customer for
connection to the customer's PBX and allocates a range of telephone numbers to
this line (or group of lines) and forwards all calls to such numbers via the trunk.
MyPBX support DID, you can configure DID in inbound route. Related settings:
DID Number, Extension, Destination.
Figure G-1
·DID Number
Define the expected DID Number if this trunk passes
Leave this field blank to match calls with any or no
provider, E1 trunks, BRI trunks or SIP trunks need to
setting.
You can also use pattern matching to match a range of
patterns may be used:
X: Any Digit from 0-9
Z: Any Digit from 1-9
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DID on incoming calls.
DID info. Only service
be configured with this
numbers. The following
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N: Any Digit from 2-9
[12345-9] : Any digit in the brackets (in this example, 1,2,3,4,5,6,7,8,9)
The ‘.’ Character will match any remaining digits. For example, 9011. will match
any phone number that starts with 9011, excluding 9011 itself.
The ‘!’ will match none remaining digits, and causes the matching process to
complete as soon as it can be determined that no other matches are possible.
·Extension
Define the extension for DID number, this field only valid when use E1 trunk for
this inbound router. You can only input number and '-' in this field, and the
format can be xxx or xxx-xxx. The count of the number must be only one or
equal the count of the DID number.
·Destination
If you don’t set the extension, you can set the destination of the call here.
Example 1:
Step1: You set the DID number (5503XXX in this example).
Step2: You choose the destination (Ring Group in this example).
The configuration of this example means when the incoming call with DID
number 5503XXX (7 digits number start with 5503) will go to the destination
Ring Group.
If you choose the destination, please leave the Extension form blank.
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Figure G-2
Example 2:
Step1: You set the DID number (6001-6099 in this example).
Step2: You set the Extension (6001-6099 in this example).
The configuration of this example means when the incoming call with DID
number 6001 to 6099 will go to the destination 6001 to 6099(number 6001 to
extension 6001, number 6002 to extension 6002).
The destination you set below will be disabled if you set the Extension.
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Figure G-3
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APPENDIX H How to Use BLF Key
to Choose the PSTN line.
MyPBX allows you to choose the specific PSTN line to make outbound call by
pressing the BLF key on the IP Phone.
Follow the steps to do the configuration with your Yealink phone
1. We want to choose pstn1 or pstn2 to call out.
Figure H-1
2. Configure the IP Phone:
Figure H-2
Test
When you press DSS Key 1/2, the phone will connect to pstn1/pstn2 line. If
pstn1/pstn2 is not busy, you will hear the dial tone. You can dial the number you
want and use this line to call out then.

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