Yeastar MyPBX Standard, MyPBX Pro User Manual

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Standard/Pro
User Manual
Version 2.16.0.58
Yeastar Technology Co., Ltd.
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Table of Contents
1. Introduction ............................................................................................ 4
1.1 Features ............................................................................................ 4
1.2 Hardware Specifications ....................................................................... 5
1.2.1 Exterior Appearance ...................................................................... 5
2. System set up ......................................................................................... 7
2.1 Connection Drawing ............................................................................ 7
2.2 Connecting Ethernet Line ..................................................................... 8
2.3 Supplying Power ................................................................................. 8
3. Managing MyPBX ..................................................................................... 9
3.1 Administrator Login ............................................................................. 9
3.2 Status Monitor .................................................................................. 10
3.2.1 Line Status ................................................................................. 10
3.3 Basic ............................................................................................... 11
3.3.1 Extension ................................................................................... 11
3.3.2 Trunk ........................................................................................ 20
3.3.3 Outbound Routes ........................................................................ 34
3.3.4 Phone Provisioning ...................................................................... 37
3.4 Inbound Call Control ......................................................................... 45
3.4.1 IVR ........................................................................................... 45
3.4.2 Queues ...................................................................................... 48
3.4.3 Custom Prompts ......................................................................... 52
3.4.4 Ring Groups ............................................................................... 52
3.4.5 Inbound Routes .......................................................................... 54
3.4.6 Blacklist ..................................................................................... 59
3.5 Internal Settings ............................................................................... 61
3.5.1 Options ...................................................................................... 61
3.5.2 Business Hours ............................................................................ 63
3.5.3 Feature Codes ............................................................................ 64
3.5.4 SIP Settings ............................................................................... 67
3.5.5 IAX Settings ............................................................................... 72
3.5.6 Voicemail Settings ....................................................................... 74
3.5.7 SMS Settings .............................................................................. 76
3.5.8 DISA ......................................................................................... 79
3.5.9 Conferencing .............................................................................. 80
3.5.10 Paging Groups .......................................................................... 81
3.5.11 DNIS Settings ........................................................................... 83
3.5.12 PIN User Settings ...................................................................... 83
3.5.13 Callback Settings ....................................................................... 84
3.5.14 Speed Dial Settings ................................................................... 86
3.5.15 Music on Hold Prompts ............................................................... 87
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3.6 Network Settings .............................................................................. 87
3.6.1 LAN Settings .............................................................................. 87
3.6.2 WAN Settings ............................................................................. 89
3.6.3 Static Route ............................................................................... 90
3.6.4 Firewall ...................................................................................... 91
3.6.5 DHCP Server .............................................................................. 95
3.6.6 VLAN Settings ............................................................................. 95
3.6.7 VPN Settings .............................................................................. 97
3.6.8 DDNS Settings ............................................................................ 98
3.7 System Settings ............................................................................... 99
3.7.1 External Storage ......................................................................... 99
3.7.2 Password Settings ....................................................................... 99
3.7.3 System Prompts ....................................................................... 100
3.7.4 Date and Time .......................................................................... 101
3.7.5 Backup and Restore................................................................... 102
3.7.6 Reset and Reboot ...................................................................... 102
3.7.7 Firmware Update....................................................................... 103
3.8 Reports ......................................................................................... 104
3.8.1 Call Logs .................................................................................. 104
3.8.2 System Logs ............................................................................. 105
3.8.3 System Info ............................................................................. 105
4. Access MRI ......................................................................................... 107
5. Use MyPBX ......................................................................................... 111
5.1 Make outbound call ......................................................................... 111
5.1.1 Sample Routing via PSTN Trunk .................................................. 112
5.1.2 Sample Routing via VoIP Trunk ................................................... 113
5.2 Incoming call ................................................................................. 114
5.2.1 Sample Routing to an IVR .......................................................... 114
APPENDIX A FAQ ..................................................................................... 117
APPENDIX B How to Configure Autobackup ................................................. 118
APPENDIX C How to Configure NAT setting ................................................. 120
APPENDIX D How to Use Auto Provision ..................................................... 122
APPENDIX E How Do I Configure Distinctive Ring Tones ............................... 125
APPENDIX F How to Use Email to SMS ....................................................... 127
APPENDIX G How to Use DID .................................................................... 129
APPENDIX H How to Use BLF Key to Choose the PSTN line. .......................... 133
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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.
Applications
1.1 Features
● Auto-provision
Firewalls
Blind Transfer
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
Distinctive Ringtone
WAN
Do Not Disturb(DND)
PPPoE
Dial by Name
Static Route
Note: the features with asterisk (*) will be supported in next version.
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1.2 Hardware Specifications
1.2.1 Exterior Appearance
A. MyPBX Standard
Front Side
Figure 1-1 MyPBX Front Panel Picture
No.
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. 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
①② ③④
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B. MyPBX Pro
Front Side
Figure 1-2 MyPBX Pro Front Panel Picture
No.
Identifying
Green LED: Indicates correct power is being supplied to the unit
Green LED: Indicates the MyPBX Pro 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. 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
①② ③④
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2. System set up
2.1 Connection Drawing
A. MyPBX Standard
Figure 2-1
VoIP Provider
PSTN
Normal
IP
601
602
603
604
FXS
LAN
Soft
Route
GSM
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B. MyPBX Pro
Figure 2-2
2.2 Connecting Ethernet Line
MyPBX provides two 10/100M Ethernet ports with RJ45 interface and LED indicator. Plug Ethernet line into MyPBXs Ethernet port, and then connect the other end of the Ethernet line with a hub, switch, router, LAN or WAN. Once connected, check the status of the LED indicator. A yellow LED indicates the port is in the connection process, and a green LED indicates the port is properly connected.
2.3 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 hasnt 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 hasnt 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
Dont 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.
6Other 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 MyPBXs call waiting function.
.DND
Dont 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:
1General
·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.
2Busy 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 <Busy Count> times as a busy signal. If you specify Busy Pattern, then MyPBX will further check the length of the tone and silence, which will further reduce the chance of a false positive disconnect.
·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
3Advanced 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
1General
·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
1General
·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 providers 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 providers 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.
·C odecs
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 providers 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 providers 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.
·C odecs
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.
1General
·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 Phones 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‟. 1General
·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.
2Key Press Events
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A list of actions that can be performed depending on the digit dialed by the user .
·K ey
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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1General
·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".
2Agents 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: 1Click 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
1General
·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.
1General
·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 extensions 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 extensions 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.
·D estination
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 extensions 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 extensions 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 ports 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.
·F ollow 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.
·D ial 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.
3Codecs 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. Its 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.
4QOS 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.
2Codecs 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: Dont 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 ports 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 OpenVPNWAN portLAN 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 hasnt 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 accounts 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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