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MyPBX Standard/GSM

Yeastar MyPBX Standard/GSM User Manual Version 2.10.0.99

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Page 1
Standard/GSM
User Manual
Version 2.10.0.99
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 ......................................................................................... 6
2.1 Connection Drawing ............................................................................ 6
2.2 Connecting Ethernet Line ..................................................................... 6
2.3 Supply i ng Pow e r ................................................................................. 6
3. M anaging MyPBX ..................................................................................... 8
3.1 Administrator Login ............................................................................. 8
3.2 Sta tus Monitor .................................................................................... 9
3.2.1 Line S ta tu s ................................................................................... 9
3.3 Basic ............................................................................................... 10
3.3.1 Extension ................................................................................... 10
3.3.1.1 Analog Extensions (FXS) ........................................................ 11
3.3.1.2 VO IP Ex te nsion ..................................................................... 13
3.3.2 Trunk ........................................................................................ 17
3.3.2.1 Analog Trunk（FXO） ............................................................. 17
3.3.2.2 GSM Trunk ........................................................................... 19
3.3.2.3 VO IP Tr un k ........................................................................... 19
3.3.2.4 Service Provider .................................................................... 21
3.3.3 Outbound Routes ........................................................................ 23
3.3.3.1 New Outbound Route ............................................................. 23
3.3.3.2 Delete Outbound Route .......................................................... 25
3.3.4 Auto Provision ............................................................................ 25
3.3.4.1 Phone .................................................................................. 25
3.3.4.2 IP Phone .............................................................................. 26
3.4 Inbound Call Control ......................................................................... 27
3.4.1 IVR ........................................................................................... 27
3.4.1.1 Cre ate IVR ............................................................................ 27
3.4.1.2 Delete IVR ............................................................................ 29
3.4.2 Queues ...................................................................................... 29
3.4.3 Custom Prompts ......................................................................... 31
3.4.4 Ring Groups ............................................................................... 33
3.4.4.1 Create Ring Group ................................................................. 33
3.4.4.2 Delete Ring Group ................................................................. 34
3.4.5 Inbound Routes .......................................................................... 34
3.4.5.1 Create Inbound Route ............................................................ 35
3.4.5.2 Delete Routes ....................................................................... 38
3.4.6 Blacklist ..................................................................................... 38
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3.4.5.1 C r eate Bla cklist ..................................................................... 38
3.5 Other Se tt in g s .................................................................................. 39
3.5.1 DISA ......................................................................................... 39
3.5.2 Conferencing .............................................................................. 40
3.5.3 Business Hours ........................................................................... 41
3.5.4 Feature Codes ............................................................................ 41
3.5.5 Pa g i ng Gro ups ............................................................................ 44
3.6 Syste m Se t ti ngs ............................................................................... 46
3.6.1 Options ...................................................................................... 46
3.6.2 Music on Hold Prompts................................................................. 47
3.6.3 SIP Se ttings ............................................................................... 47
3.6.4 IAX Settings ............................................................................... 50
3.6.5 Voic email Settings ....................................................................... 52
3.6.6 Network Setting .......................................................................... 54
3.6.7 DHCP Server .............................................................................. 55
3.6.8 Exte rnal Storage ......................................................................... 56
3.6.9 Password Setting s ....................................................................... 57
3.6.10 Date and Time .......................................................................... 58
3.6.11 Backup and Restore ................................................................... 59
3.6.12 Reset and Reboot ...................................................................... 60
3.6.13 Firmw are Update ....................................................................... 61
3.7 Reports ........................................................................................... 62
3.7.1 Call Logs .................................................................................... 62
3.7.2 System Info ............................................................................... 63
4. Access MRI ........................................................................................... 65
5. Use MyPBX ........................................................................................... 71
5.1 Make out b ound call ........................................................................... 71
5.1.1 Sample Routing via PSTN Tr un k .................................................... 72
5.2 Incoming call.................................................................................... 75
5.2.1 Sample Routing to an IVR ............................................................ 75
APPENDIX A FAQ .................................................................................... 77
APPENDIX B How to Configure Autobackup ................................................ 78
APPENDIX C How t o Configure NAT setting ................................................ 80
APPENDIX D How to Use Auto Provision .................................................... 83
APPENDIX E How do I configure distinctive ring tones ................................. 86
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1. Introduction
MyPBX —An IP-PBX for Small Businesses/Home Office
MyPBX is a standalone embedded hybrid PBX for small businesses and remote branch offices of larger organizations (2-100 users per site). MyPBX also offers a hybrid solution (a combination of VoIP applications using legacy telecom equipment) alternative for enter prises who are not yet ready to migrate to a complete VoIP solution.
Applications
1.1 Features
● Auto-provision ● Do Not Disturb(DND)
● Blind Transfer ● Firewalls*
● BLF Support ● Follow me
● Blacklist ● Interactive Voice Response (IVR)
● Call Det ai l Records(CDR) ● Intercom / Zone Intercom
● Call Forward ● Music On Hold
● Call Parking ● Music On Transfer
● Call Recording ● Paging / Zone Paging
● Call Pickup ● Queue
● Call Routing ● Ring Gro up
● Call Transfer ● Route by Caller ID
● Call Waiting ● Skype Integration (Skype for SIP)
● Caller ID ● Three-way Calling
● Conference ● T.38
● Define Office T ime ● Voicemail
● Direct Inward System Access(DISA) ● Voicema il to e ma i l
● DIDs ● Voicemail forwarding
● Distinctive Ringtone
Note: the features with indication (*) will be supported in next version.
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1.2 Hardware Specifications
1.2.1 Exterior Appearance
Front Side
Figure 1-1 MyPBX Front Panel Picture
No. Identifying
①
Green LED：Indicates correct power is being supplied to the unit
②
Green LED：Indicates the MyPBX is fully functional.
③
Green LED：Indicates stable Ethernet0 connection
④
Green LED：Indicates stable Ethernet1 connection
⑤
Red LED：Indicates presence of an FXO/GSM module.
Green LED：Indicates presence of an FXS module .
LED Blinking- Red blinking：N o connection betwe en F XO p or t 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 blinking：FXS port is ringing LED Dual - Green and Red blinking：FXS port is in use
①② ③④ ⑤
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2. System set up
2.1 Connection Drawing
2.2 Connec t ing Ethe rn et Lin e
MyPBX provides two 10/100M Ethernet ports with RJ45 interface and LED indicator. Plug Ethernet line into MyPBX’s Ethernet port, and then connect the other end of the Ethernet line with a hub, switch, router, LAN or WAN. Once connected, check the status of the LED indicator. A yellow LED indicates the port is in the connection process, and a green LED indicates the port is properly connected.
2.3 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, 3A
PSTN
PSTN
Normal Phone
IP Phone
601
602
603
604
FXS Lines LAN
IP Provider
Soft Phone
Router
GSM
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Please follow the steps below to connect the MyPBX unit to a power outlet:
1. Connect the small end of the power c able to t he power input port on the MyPBX back panel, and plug the other end of the cable in to a 100VA C 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, pleas e use the default settings below: IP Address: http://192.168.5.150 Username: admin Password: password
Figure 3-1
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3.2 Status Mon ito r
3.2.1 Line Status
Figure 3-2
MyPBX Status Description:
Extensions:
1)
: Extension is unregistered
2)
: Extension is idle
3）
: Extension is ringing
4）
: Extension is busy in a call
Trunks:
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VOIP Trunk:
Status
Unregistered: Trunk registration failed.
Registered: Succeed registration, trunk is ready for use. Request Send: Registering. Waiting: Waiting for authentication.
FXO Trunk:
Status
Idle: The port is idle.
Busy: The port is in use. Disconnected: The port hasn’t connected the PSTN line.
More detail message, please refer to the LED identifying of front panel.
GSM Trunk:
Status
Idle: The port is idle.
Busy: The port is in use.
Signal
: No signal. : Poor. : Average. : Good. : Excellent.
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
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.
·Name A character-based name for this user, i.e. ‘Bob Jones’
·Caller ID The Caller ID (CID) string will be used when t his user calls another interna l user.
2) Voicemail
·Voicemail Access PIN #
Voicemail Password for this extension, i.e. ‘1234’.
·Send Voicemail to Email Address
This option defines whether or not voicemails/Fax are sent to the Email address as an attachment. Note: Please ensure that all voicemail settings are properly configured on the System Settings -> Voicema il Sett ings pa g e before using this feature.
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3) 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 ev ent 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 i ncrements.
4) Group
·Pickup Group If this extension belongs to a pickup group, any calls t hat 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.
5) Follow me (Call Forwarding) This function set s inbound call forwa rding on an extension. An administrator can configure Follow Me for this extension.
6）Other Options
.Call Waiting
Check this option if the extension should have Call Waiting capability. If this option is checked, the ‘When busy’ f ollow me op tions will not be available. .User Web Inter face Check this option to allow the user to login to the MyPBX User Web interfa ce, which can be used to acce ss voice ma il a nd e xt e nsion recordings. Users may login to the 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 custo m the ring time. Tone wil l stop over the time defined.
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Figure 3-4
3.3.1.2 VOIP Extension
A VOIP extension is an 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
1) General
·Type Extension type : SIP or IAX. SIP – Identifies whether the extension sends and receives calls using the VoIP protocol SI P. IAX - Identifies whether the extension sends and receives calls using the VoIP protocol IA X .
·Extension The numbered extension, i.e. 1234, that will be associated with this particular
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User / Phone.
·Password The password for this extension, Ex: ‘12t3f6’
·Name A character-b a s ed name for this user, EX : ‘Bob Jones’
·Caller ID The Caller ID will be used when this user calls a not he r int e rna l extension.
2) Voicemail
·Enable Voicemail
Check this box if t he use r should hae a voicemail account.
·Voicemail Access PIN #
The voicemail Password for this extension, i.e. ‘1234’.
3) Mail Setting This option def i ne s whe t he r or n ot voicemails a re sent to an Email Address as attachment.
·Enable Send Voicemail
Once enabled, the voicemail or Fax will be sent to email as an attachment.
·Email Address
Email address use to receive the voicemail or Fax. Note: Please ensure that the section 'SMTP Settings For Voicemail'(in the 'Voicemail Settings') have been properly configured before using this feature.
4) VoIP Settings
·NAT This setting should be used w hen t he sy st em is usi ng a public IP address, communicating with d evic es hidd en b ehind a NAT dev ice (such a s a b roadb and router). If you ha ve one -way audio problems, you usually have problems with your NAT config ura t i on o r y our f ire wall's support of SIP and/or RTP ports.
·DTMF Mode – RFC2833, Info, Inband, Auto.
5) IP Restriction
·Enable IP Restriction Check this option to enhance the VoIP security for MyPBX. If this option is enabled, only the permitted IP or Subnet mask will be able to register this extension numb e r. I n t hi s way, the VoIP securit y w il l b e e nha nc ed.
·Permitted ‘IP address/Subnet mask’ The input forma t shoul d b e ‘IP address’+’/’+’Subnet mask’. e.g."192.168.5.100/255.255.255.255" me a ns only the device whose IP address is 192.168.5.100 i s a llowed t o register this extension number. e.g."192.168.5.0/255.255.255.0" means onl y the device whose IP add ress is
192.168.5.XXX is allowed t o regist e r t hi s e xt ension number.
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6) Group
·Pickup Group If this extensi on belongs to a pickup group , any calls that ring this extensi on 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 set s inbound call forwa rding 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 opt ions will not be availab le. .User Web Inter face/Web Access Check this option to allow the user to login to the MyPBX User Web interfa ce, which can be used to acce ss voice ma il a nd e xt e nsion recordings. Users may login to the 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 custom the ri ng t ime . Tone wi ll st op ove r t he time defined.
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Figure 3-5
2. Add Multiple Extensions Go to Extensions VOIP Extensions Add Multiple Extensions.
1) Choose the number of extensions that you would like to create.
2) Choose the type of extension that you would like to create.
3) Fill in the sta rt ing ex t e nsion number.
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Figure 3-6
3. Edit VOIP Extension Click ‘Edit’ on VOIP Extension administration page or click ‘Modify Selected Extensions’ to edit extension.
3.3.2 Trunk
There are three types of trunks: Analog trunks (FXO), VOIP and Service Provider trunks.
Figure 3-7
3.3.2.1 Analog Trunk（FXO）
1. Edit Analog Trunk（FXO）
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On the analog Trunk administration page, click ‘Edit’ on the selected trunk and modify its properties in the popup window:
1）General
·Trunk Name A unique label used to identify this trunk when listed in outbound rules, incoming rules, etc. Ex: 'Port 5'
·Volume Setting
Used to modify the volum e level of this tr unk. Normally, this settin g does n ot need to be changed.
2）Advanced Options
·Busy Detection Busy Detection is used to detect far end hang-up or for detecting a busy signal. Select “Yes” to turn this feature on.
·Busy Count If Busy Detect ion is enab led, it is also possib le to sp ecify how ma ny busy to nes to wait for before disconnecting. The default is 4, but better results can be achieved if set to 6 or even 8. Remember, the higher the number, the more time required to release a channel. A higher se tti ng lowers the probability that you will encounter random ha ng -ups
·Busy Pattern If Busy Detection is enabled, it is also possible to specify the cadence of your busy signal. In many countries, it is 500msec on, 500msec off. Without Busy Pattern specified, MyPBX will accept any regular sound-silence pattern that repeats <Busy Co unt> times as a busy signal. If you spe cify Busy Patte rn, then MyPBX will further check the length of the tone and silence, which will fur ther reduce the chance of a false positive disconnect.
·Caller ID Start This option allows you to define the start of a Caller I D signal: Ring: Start when a ring is received (Caller ID Signaling: Bell_USA, DTMF). Polarity: Start when a polarity reversal is started (Caller ID Signaling: V23_UK,V23_JP,DTMF). Before Ring: Start before a ring is received (Caller ID Signaling: DTMF).
·Caller ID Signaling
This option defines the type of Caller ID signaling to use. It can be set to one of the following: Bell: bell202 as used in the United States v23_UK: Used in the UK v23_JP: Used in Japan DTMF: Used in Denmark, Sweden, and Holland
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Figure 3-8
3.3.2.2 GSM Trunk
1. Edit GSM Trunk
1）General
·Trunk Name A unique label used to identify this trunk when listed in outbound rules, incoming rules, etc. Ex: 'gsm1'
·Volume Setting
Used to modify the volum e level of this tr unk. Normally, this settin g does n ot need to be changed.
Figure 3-9
3.3.2.3 VOIP Trunk
1. Add SIP Trunk
Input correct SIP information (provided by VOIP provider). Inaccurate information wi ll p re ve nt the trunk from registering.
·Type SIP – Identifie s whet he r t he t runk se nd s a nd re ce iv e s ca l ls us ing t h e V oI P protocol SI P.
·Provider Name A unique label to help you identify this trunk when listed in outbound rules, incoming rules e t c. Ex: 'yeastar'.
·Hostname/IP
Service provider’s hostname or IP address.5060 is th e standard port number used by SIP protocol. Don’t change this part if it is not required.
.Domain
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VoIP provider’s serv e r d oma i n name.
·Username
Username of SIP account. Used for SIP trunk registration.
·Password
Password of SIP account.
.Authorization name
Used for SIP authentication. Lea v e this blank if not req ui red.
.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 ma ximum n umber of outbound c hannel s (simulta ne ous cal ls) tha t can be used on this tr unk. Inbo und call s are no t counted again st the max imum. Leave blank to specify no maximum.
Figure 3-10
2. Add IAX Trunk
Input correct IAX information (provided by VOIP provider). Inaccurate information wi ll p re ve nt the trunk from registering.
·Type IAX – Identifies whether the trunk sends and receives calls by using the VoIP protocol IA X .
·Provider Name A unique label to help you identify this trunk when listed in outbound rules, incoming rules e t c. Ex: 'yeastar2'.
·Hostname/IP
Service provider’s hostname or IP address. 4569 is the standard port number used by IAX protocol. Don’t change this part if it is not required.
·Username
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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 ma ximum n umber of outbound c hannel s (simulta ne ous cal ls) tha t can be used on t his trunk . Inbo und calls a re not counted a gainst t he maximu m. Leave blank to specify no maximum.
Figure 3-11
3.3.2.4 Service Provider
1. Add Service Provider
To Create the Service provider definition y ou need to com plete the following VoIP fields.
·Type SIP or IAX SIP – Identifies whether the trun k sends and receiv es calls by using the VoIP protocol SI P. IAX - Identifies whether the trunk s ends and receives calls by using the VoIP protocol IA X .
·Provider Name
A unique label would help to you identi f y this trunk. Ex: ' Provider2'.
·Hostname/IP
Service provider’s hostname or IP address. Note: 5060 is the standard port number u sed by SIP protocol, 4569 is the standard port number used by IAX protocol. Don’t change this part if it is not required.
·Maximum Channels
Controls the max imum n umber of outbound ch annels (simulta ne ous call s) tha t
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can be used on t his trunk . Inbo und calls a re no t counted a gainst the maximu m. Leave blank to specify no maximum.
.DOD Setting
1) DOD – Direct Outward Dialing Number.
2) Bind Extensi on.
Figure 3-12
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3.3.3 Outbound Routes
Outbound routi ng defines how outgoing ca lls are p rocessed thro ugh the t runks.
Figure 3-13
3.3.3.1 New Outbound Route Click ‘New Outbound Route’ and fill in the corresponding information in the popup window. 1）General
·Route Name Name of this Outbound Route. Ex: 'L oca l ' or 'Long Distance' e t c.
·Dial Pattern Outbound calls t hat mat ch this dia l 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 t hat starts wit h 90 11 , e xcluding 9011 itself . The ‘!’ will match any 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 numbe r.
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Example 2: 1NXXNXXXXX will match a phone number starting with a 1, followed by a 3-digit area code, and then 6 digit number.
·Strip digits fro m fron t
Allows th e user to specify the n umber o f digi ts that wi ll be stripped from th e front of the p hone numbe r bef ore the call is place d. F or exa mple , if users must press 0 befor e di alin g a pho ne n umbe r, o ne digit sho uld be str ipped from the dial string before the call is placed.
·Prepend these digits before dialing
These digits will be prepend ed to the p hone nu mber 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 ph one numbers bef ore call s are p lace d. When usi ng a nalog trunks, a 'w' character may also be prepended to provide a slight delay before dialing.
·Member Extensions
Defines the e x t en sions t hat will be permit t ed to use this outbound r out e .
·Member Trunks
Defines the t runk s t hat can be used for t hi s out b ound rout e .
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Figure 3-14
3.3.3.2 Delete Outbound Route
Click ‘Delete’ on selected trunk to delete it.
3.3.4 Auto Provision
3.3.4.1 Phone
Click ‘Create New Phone’ and f ill in the corresponding in format ion in the popup window. 1）General
·MAC
IP Phone’s MAC address.
·Name
Name of this Phone.
·Call Waiting
This call feature allows your phone to accept other incoming calls to an extension already in an active call.
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·Key As Send
·Line
Extension: Selected the extension number for IP Phone. Label: It is shown on the LCD for users t o i d e nt ify the account. Line Active: You can choose o n/off to enable/disable the account respectively.
Figure 3-15
3.3.4.2 IP Phone
Click ‘Scan Yealink Phone’, System will scan Yealink’s IP phone in the local network, and list IP phone’s MAC and IP address in this page.
Click ‘MAC address’, and fill in the corresponding information in the popup window. Like 3.3.4.1
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3.4 Inbound Ca ll Contro l
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.
Figure 3-16
3.4.1.1 Create IVR
Click ‘Create New IVR’. 1）General
·Number
MyPBX treats IVR as an extension; you can dial this extension numbe r 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
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The number of times that the selected IVR prompt will be played.
·Key Timeout
Wait for the user to enter a new extension for a specified number of seconds.
·Allow Dialing of Other Extensions
Allow the caller to dial other extensions other than the ones explicitly defined.
2）Key Press Events A list of actions that can be performed depending on the digit dialed by the user.
·Time Out
Defines the time out act ion. A timeout occurs af ter t he IVR prompt ha s finished playing for the number of t imes 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 3-17
3.4.1.2 Delete IVR
Select the item that you would like to remove and press the ‘Delete’ button.
3.4.2 Queues
Call queues allow calls to be sequenced to one or more agents. 1）General
·Queue Name
A name for the Que ue.
·Queue Number
Use this number t o dial into the queue, or t ransfer calle rs to this number to put them into the q ue ue .
·Queue Password
You can requi re agent s to enter a pa ssword bef ore they ca n login to thi s queue.
·Queue Agent Timeout
The number of second s an agent's phone ca n ring before we consider it a timeout.
·Queue Max Wait Time
The maximum numb e r of se conds a calle r ca n w a it i n a que ue b efore b e ing pulled out. (0 for unlimited).
·Queue Ring Strategy
This option sets t he Ring i ng S t ra t e g y f or t his Queue. The options are RingAll: Ring All available Agents simultaneously until one answers. LeastRecent: Ring the Agent which was least recently called. FewestCalls: Ring the Agent with the fewest completed calls. Random: Ring a Random Agent. RRmemory: Round Robin with Memory, Remembers where it left off in the last ring pass".
2）Agents This selection shows all users. Selecting a user here makes them a agent of the current queue.
3) Caller Position Announceme nt s
·Announce Position
Announce posit ion of caller in t he queue
·Announce Hold Time
Enabling thi s op t ion causes MyPBX to anno unce the hold t ime t o t he ca l le r periodically based on the frequency t imer. Eit her yes or no; hold time will n ot be announced if <1 minute.
·Frequency
How often to announce queue posit i on a nd e st i ma t e d hold time.
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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 (0 to Disable A nnouncements).
5) Events
If a caller presses the key while wai ting in the que ue, thi s sett ing se lects 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 'M usi c on Hold ' Cl a ss for this Queue .
·Leave When Empty
This option cont rols w he t her callers alre a d y on hold a re forced out of a que ue that has no a g e nt s. T he re are two options. Yes: Callers are forced out of a queue when no agents are logg ed in. No: Callers will remain in a queue with no agents.
·Join Empty
This option cont rols w he t her callers can j oin a call queue tha t has no a g e nt s . There are two options, Yes: Callers can join a call queue with no ag ents or only una v a il a b l e agents No: Callers cann ot joi n a q ue ue wi t h no a g e nt s The default option is No.
·Agent Announcement
Announceme nt p l a y ed to the Agent prior t o b ridgi ng i n t he ca l le r.
·Join Announcement
Announceme nt p l a y ed to callers once pri or to j oinin g t he q ue ue .
·Retry
The number of second s we wait before trying all the phones aga in. Choosing 'No Retry' will exit the Queue a nd go to the fa il-over destination a s soon as the first attempted agent times-out, additional agents will not be attempted.
·Wrap-up time
How many seconds after the completion of a c all an Agent will have before the
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Queue can ring the m wi t h a ne w ca ll . T he d e faul t is 30.
Figure 3-18
3.4.3 Custom Prompts
1. Record new Prompt
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The administr ator can use this screen to record custom prompts by doing the following: 1）Click ‘Record New Custom Prompt’
2) Input the desired file name on the popup window and choose a n extension to
call for recording (such a s 500).
3) Click ‘Record’. The selected extension will ring and you can pick up t he phone
to start recording.
Figure 3-19
2. Upload Prompt
The administrator can also upload prompts by doing the following:
1)）Click ‘Upload Prompt’.
2) Click ‘Browse’ t o cho ose t he desired prompt.
3) Click ‘Upload’ to upload the selected prompt.
Figure 3-20
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3.4.4 Ring Groups
Figure 3-21
3.4.4.1 Create Ring Group
Click ‘New Ring Group’ to enter into the Manage Ring Groups page
1）General
·Ring Group Name
This option defines a name for this group, i.e. ‘Sales’. ‘Ri ng Gr oup N ame’ is a label to help you identify this group in t he g roup list .
·Ring Group Number
This option defines the numbered extension that can be dialed to reach this group.
·Strategy
This option sets t he Ring ing Strat egy for t his Group. The opt ions are as follows:
1. Ring All Simultaneously: Ring all available Extensions simultaneously.
2. Ring Sequentially: Ring each e x t en sion in the group one at a time.
·Seconds to ring each member
Set the numbe r of seconds to ring this group bef ore routing the call according to the ‘Destination if No Answer’ settings.
2) Ring Group Members
An extension can be made a member of this ring group by moving it into the ‘Selected’ box.
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3) Destination If No Answer
When all members on t his group fail to answer the call, syst e m will handle the call according to the selected destination.
Figure 3-22
3.4.4.2 Delete Ring Group
Select the item you’d like to remove and press the ‘Delete’ button.
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 3-23
3.4.5.1 Create Inbound Route
Click ‘New Inbound Rout e ’ to enter to the Manage Inbound Routes page. 1）General
·Route Name
A name for this inbound route. Ex: 'pstncallin' etc.
·DID Number
Define the expected D ID 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 or E1 tru nk s ne e d t o b e conf igured with this se t t i ng . You can also use pa ttern ma tching to mat ch a range of numbe rs. 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 t hat starts wit h 90 11 , e xcl ud i ng 90 11 i t se lf. The ‘!’ will match any 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.
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·Caller ID Number
Define the Caller ID N u mber to be match ed on incoming calls. Leave this field blank to mat ch any or no DID info. You can also use a pattern match (e.g. 2[345] X) to mat ch 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 t hat starts wit h 90 11 , e xcl ud i ng 90 11 i t se lf. The ‘!’ will match any 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 exa mple, if you configure the distinctive ringing for custom ring tone to ‘Family’, the ring tone will b e played if the phone receives the incoming call.
How do I configure distinctive ring tones? Please see
APPENDIX E.
2）Member Trunks This area al lows you to select which t runks will be member trunks f or this route. To make a trunk a member of this route, please move it to the 'Selected' box. 3）During Office Hours Y ou can choose how to rout e incoming calls during office hours. 4）Outside Office Hours Y ou can choose how to rout e incoming calls outside office hours.
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Figure 3-24
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3.4.5.2 Delete Routes
Select the item you would like to rem ove and click ‘Delete’ key.
3.4.6 Blacklist
Number Blacklist is used to block an incoming call you do not want to answer. If the incoming ca ll number is registered in the number blacklist, the caller will hear the following: “The number you have dialed is not in service. Please check the number and t ry again”. The system will then disconnect the call.
Figure 3-25
3.4.5.1 Create Blacklist
Click ‘New Blacklist’ to create a new number blacklist.
Figure 3-26
·Blacklist number
Enter the number y ou would like to b l ock .
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3.5 Other Settings
3.5.1 DISA
DISA (Direct Inwa rd System Access) al lows someone calling in from outsid e the telephone swit ch (PBX) to obtai n an ‘internal’ system dial tone and make calls as if they were using o ne of the extensi ons 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 3-27
To add a new DISA application, click the New DISA button.
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Figure 3-28
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 amo unt of time the syste m will wait before hanging up t he call if the user has dialed an incomplete or invalid number. Default is 10 seconds.
·Digit Timeout
The maximum amount of time permitted bet ween each digit when the user i s dialing a n e xt e nsi on numb e r. Default i s 5 se cond s .
2) Member Outbound Routes
Used to set the outbound routes that can be accessed from this DISA.
3.5.2 Conferencing
·Extension
This is the number dialed to rea ch this Conference Room.
·Pin #
Set a PIN # that must be entered in order to access this conference room (i.e.
1234).
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Figure 3-29
3.5.3 Business Hours
Y ou ca n s etup the business hours here.
Figure 3-30
3.5.4 Feature Codes
1. General
·One Touch Record
A user may initiate or stop call reco rding by dialing *1 during a c all. (*1 is default setting)
·Extension for Checking Voicemail
Users can check their Voicemail by dialing *2 on their phone (*2 is default setting).
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·Attended Transfer
Users may transfe r an incoming call b y dialing *3 on their phone (*3 is defa ult setting).
·Blind Transfer
Users may blind transfer an incoming call by dialing *03 on their phone (*03 is default setting).
·Call Pickup
Users may pic k up an incoming call by dialing *4 on their phone (*4 is default setting)
·Extension Pickup
Users may pick up a specific extension’s incom ing 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 i nterco m mode. For instance setting this value to *5 would allow you to initiate an intercom call with extension 501 by dialing *550 1.
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 exte nsion.
·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 def aults. The call forwarding settings will be configured as follows:
Always forward: Disabled Busy forward to Voicemail: Enabled No answer fo rward to Voicemail: Enabled Do not disturb: Disabled
·Enable Forward All Calls
Users may enable always forward by calling *71 on their pho ne (* 71 is default setting)
·Disable Forward All Calls
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Users may disable always forward by calling *071 on their phone (*071 is default setting)
·Enable Forward When Busy
Users may enable bu sy forw ard by dialing *72 on their phon e (* 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 forw ard by calling *73 on their phone (*73 is default setting)
·Disable Forward No Answer
Users may di s abl e 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 p hone numb e r to forward a l l 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 exte nsion 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 e n able 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 setting)
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Figure 3-31
3.5.5 Paging Groups
Paging is used to make a n announcement ov er the s peakerp hone to a phone or group of pho nes. Targete d phones will n ot ring, b ut instead answer immediately into speake rphone mode. Plea se n ote that this se ction is fo r conf iguring pa ging groups. If you would l ike to configure Int ercom setting s, please open th e Other Settings -> Feature Codes screen. This feature is sup p ort e d b y the f oll owing S I P p hone s: 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 3-32
·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 3-33
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3.6 System Settings
3.6.1 Options
1) General
·Ring Timeout
Number of seconds to ring a device before handling the call as per the extension’s Follow Me settings. Default value is 30s.
.MAX call duration The absolute maximum amount of time permitted for a call. A setting of 0 disables the timeout. Default value is 6000s.
·Music on hold
Used to set hold music f or the sy ste m.
·Tone Region
Please select your country or neares t neighb oring country to enable the default dial tone, b usy t one, a nd ri ng t one f or your re g ion.
Note: please reboot the system to take it effect.
·HTTP bind port/Web Access Port
Port to use for HTTP sessions. De f a ult : 80
2) Extension Preferences
·User Extensions
The default va lue is 500 to 616
·Paging Group Extensions
The default value is 620 to 629
·Ring Group Extensions
The default value is 630 to 639
·Conference Extensions
The default va lue 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 3-34
3.6.2 Music on Hold Prompts
The admi nistrator c an 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, Mon o, 1kb/sec
Figure 3-35
3.6.3 SIP Settings
1) General
·UDP Port
Port use for sip registrations, Default is 5060.
·RTP Port Start
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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 durat ion (in seconds) of a SIP registration. Default is 3600 se conds.
·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.
·Registerattempts
The number of SIP REGISTER messages to send to a SIP Registrar before giving up. Default is 0 (no limit).
·Registertimeout
Number of seconds to wait f or a response from a SIP Registrar before timed out. Default is 20 seconds.
·Videosupport
Support for SIP video or no. Default is yes.
2) NAT
Note: onfiguration of this section is only required when using remote extensions.
·External IP Address
The IP address tha t will be associated wit h 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 y our ISP for more i n f ormation.
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·External Refresh Interval
If an ext ernal host has been sup plied, you may specify how often the system will perform a DNS query on this host. This value is specif ie d in se conds.
·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 con f l ocal netw ork . Please refe r t o RF C19 18 f or more informa t i on.
·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 acco rd ing t o RFC35 81 . Never = Never attempt NAT mode or RFC3581 support. Route = Use NAT but do not include report 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 r oute packets to each other directly, bypa ssing the system. It is not always possible for the system to negotiate endpoint-to-endpoint media routing.
3）Codecs A codec is a compression or decompression algorithm that used in the transmiss ion of voice packe ts over a ne twork or the Internet. u-law: A PSTN standard codec, used in North America, that provides very good voice quality and consumes 64kbit/s in eac h d irec tion ( rec eiving an d transmitting) of a VoIP call. a-law: A PSTN standard codec, used outside of North America, that provides very goo d voice quality a nd consumes 64kbit/s in each d irectio n (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 transmitt ing) of a VoIP call. GSM is supported by man y VoIP ph o nes.
SPEEX:
Speex is an Open Source/Free Software patent-free audio
compress ion for mat des igned for speech. The Speex Pro ject a ims to low er
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the barrier of entry for voice applications by providing a free alternative to expensive pr oprietar y speech cod ecs. Moreover, Sp eex is well-adapted to Internet applica tions and provides useful features that are not present in most other codecs. G.726: A PS TN codec, u sed worl dwide, t hat provi des good v oice qual ity and consumes 32kbit/s in each dir ec t ion ( r ec eiving and tra n sm it t ing ) o f a VoIP call. G.726 is supported by some VoIP phones.
ADPCM, G.72 9, H2 61 , H26 3, H26 3p , H264. Note: If you would like to use G.729, please enter your license.
Figure 3-36
3.6.4 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 durat ion (in seconds) of a IAX2 registration. Default is 1200 seconds.
2）Codecs A codec is a compression or decompression algorithm that used in the transmiss ion of voice packe ts over a ne twork or the Internet. u-law: A PSTN standard codec, used in North America, that provides very good voice quality and co nsumes 64kbit/s in each d irec tion (re ceiving an d transmitting) of a VoIP call. a-law: A PSTN standard codec, used outside of North America, that provides very goo d voice quality a nd consumes 64kbit/s in each d irectio n (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 transmitt ing) of a VoIP call. GSM is supported by man y VoIP ph o nes.
SPEEX:
Speex is an Open Source/Free Software patent-free audio
compress ion for mat des igned for speech. The Speex Pro ject a ims to low er the barrier of entry for voice applications by providing a free alternative to expensive pr oprietar y speech cod ecs. Moreover, Sp eex is well-adapted to Internet applications and provides useful features that are not present in most other codecs. G.726: A PST N codec, use d world wide, that provide s good voi ce qual ity and consumes 32kbit/s in each dir ec t ion ( r ec eiving and tra n sm it t ing ) o f a VoIP call. G.726 is support ed by some VoIP phones.
ADPCM, G.72 9, H2 61 , H26 3, H26 3p , H264. Note: If you would like to use G.729, please enter your license.
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Figure 3-37
3.6.5 Voicemail Settings
1) General Voicemail Settin g s
·Max Messages per Folder
Sets the ma ximum number of messages t hat 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
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If this option is set, the caller will be prompted to press 5 before leaving a message.
·Announce Message Caller ID
If this option is enabled, the Caller ID of the pa rty 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 beg ins 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
The IP addre ss or hostname of an SM TP server t hat the MyPBX will connect to in order to send voicemail messages via email, i.e. mail.yourcompa ny.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 t o 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 whethe r t he se t up i s OK .
1) If the test is successful, you can use the email safely.
2) If test failed, please c heck the above informatio n is correct or network is
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proper.
Figure 3-38
3.6.6 Network Setting
·DHCP
If this option is set, MyPB X will use D HCP to get an avail able I P address f rom your local network. Not recommended.
·Enable SSH
This is the advance way to access the devi ce, you can use the putty software to access the device. In the SSH access, you can do more advance se t t ing and debug.
·Port: the default is 8022,
·Hostname
Set the host name for MyPBX.
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·IP Address
Set the IP Address for MyPBX.
·Subnet Mask
Set the subnet ma sk for MyPBX.
·Gateway
Set the gateway for MyPBX.
·Primary DNS
Set the primary DNS for MyPBX.
·Secondary DNS
Set the secondary DNS for MyPBX.
Figure 3-39
3.6.7 DHCP Server
Dynamic Host Configu ratio n Protoc ol (DHC P) is a netw ork protoc ol tha t ena bles a server to automatic ally assign an IP address to a computer from a defin ed range of numbers (i.e., a scope) configured for a given network.
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Figure 3-40
3.6.8 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 c on figu r e d, MyPBX will move all qualified files to a Windows PC every 30 minutes. For the voicemail files and recoding files, they mu st be created before Auto-Backup has been conf igured.
How to configure Ext e rnal Storage , please see.
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Figure 3-41
3.6.9 Password Settings
The default passw ord is ‘password’. T o ch ange 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 3-42
3.6.10 Date and Time
Set the date and time for MyPBX.
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Figure 3-43
3.6.11 Backup and Restore
Backup / Restore MyPBX’s Configuration.
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Figure 3-44
3.6.12 Reset and Reboot
·Reboot System
Warning: Rebooting the system will termina t e a ll active calls!
·Reset to Factory Defaults
Warning: A factory reset will erase all configuration data on the system.
Please do not tu rn off t he system u ntil the RUN lig ht begi ns blinking. Any power interruption du ring this time coul d ca use d amage to the s ystem.
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Figure 3-45
3.6.13 Firmware Update
Upgrading of the firmware is possible through the Administrator web interface using a TFT P Se rv e r or an HTTP URL. Enter your TFTP Se rver IP a ddress and f irmware file location, then click start to update the firmware.
Note：
1. If enabled ‘Reset configs’, System will restore to factory default settings.
2. When update the firmware, please don’t turn off the power.
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Figure 3-46
3.7 Reports
3.7.1 Call Logs
The call Log captures all call details, including call time, caller number, callee number, call type, call duration, etc. An administrator ca n se a rch and filter call data by call time, call type and call duration and can also export CDR data to a CSV file.
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Figure 3-47
3.7.2 System Info
General:
Info rm ation about hardware version, firmware vers ion and system uptime.
Network: Information about hostname, MAC address, IP address, subnet mask, gateway, Primary DNS and Secondary DNS.
Disk Usage: Disk usage information.
Memory Usage: Memory usage inf o rm ation.
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Figure 3-48
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4. Access MRI
MRI (MyPBX Recording Interface).Users may access MRI by logg ing into the MRI web interface with their username (extension number) and voicemail password.
4.1 Allow users to acce ss MRI
The extension’s ‘User Web Interface’ option must be checked before the associated user can log int o MRI .
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Figure 4-1
4.2 User login
Users can access the MRI web interface by navigating to the MyPBX IP address using a web browser. If you are unsure of this address, please contact your network administrator.
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Figure 4-2
4.3 Voicemail
Users can check, delete, move and download voicemail files here.
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Figure 4-3
4.4 Record
Users can play, delete and download recorded files here.
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Figure 4-4
4.5 Settings
Voicemail password: new voicemail box password. Enter again t o confirm: confirm new voicemail box password. Send Voicemail To Email Address: Set the email address that voicemail messages will be sent to.
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Figure 4-5
4.6 Follow me
Set the mode of call forwarding when there is an incoming call to this extension.
Figure 4-6
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5. Use MyPBX
5.1 Make outbound call
To make an outbound call, we need to add trunk/service provider first. There are two types of service provider:
·Analog Trunk Provider: FXO ports of MyPBX, connected to a local PSTN
·SIP Trunk Provider: Connected to remote SIP VOIP service provider server
What are FXO and FXS? FXS (Foreign exchange Station) is an interface which drives an analog telephone or FAX machine. FXS in terfaces deliver power, provide r inging, and use FXO signaling. FXS in terfaces are what al lo w you to hook telephones and other analog devices to your PBX FXO (Foreign exchange Office) is an interface that connects t o a phone li ne to supply your PBX with access t o a publi c telephone network. FXO inte rface s use FXS signaling. FXO interfaces allow you to connect your PBX to real analog phone lines.
PSTN
PSTN
Normal Phone
IP Phone
601
602
603
604
FXS Lines LAN
IP Provider
Soft Phone
Router
GSM
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5.1.1 Sample Routing via PSTN Trunk
Let’s route all ins ide extens ions through an ana log trunk by dialing 9. In Outbound Routes, add a new outbound route as below.
Figure 5-1
As we can see from the outbound route of ‘pstnout’, all phone numbers starting with 9 will have their first digit stripped off (digit 9) and will be sent to the PSTN (port 1-2).
After we have configured the above, we can dial 9 + local number to dial out via a PSTN line.
Note: Setting number pre fix to wil d card X and setting Strip to 0 digits from
the front will allow all calls to go through this outbound route.
5.1.2 Sample Routing via a VoIP Trunk
Let’s configure all inside extensions to dial ‘0’ through the VoIP service provider.
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1. Add VoIP service provider
Before we do add this, please m ake sure you have a VoIP service provider account. TrunksSIP Trunk Add SIP Trunk Enter your account information on this page, and click Save.
Figure 5-2
2. Add Outbound Routes
As we can see from the Outbound Route o f ‘voipout’, all phone numbers starting with 0 will have their first digit stripped off (digit 0) and will be sent to the SIP service provider.
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Figure 5-3
Now that we have adde d two outbound dialing rule s, any call sta rting with 9 will be ro uted to the PSTN , and any number starting with 0 wi ll be rou ted to the SIP provider.
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5.2 Incoming call
5.2.1 Samp l e Routing to an IVR
Let’s configure an incoming ca ll to route to the IVR. In the IVR itsel f, let’s configure digit 0 to route the call to extension 500, and digit 1 to route the call to extension 501.
1. Add IVR
To add a new IVR, go to IVR Create New IVR
Figure 5-4
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2. Add Inbound Routes
As we can see from the I nbound Route of ‘allin’, all incoming calls will be sent to the IVR.
Figure 5-5
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APPENDIX A FAQ
Q1. How to Register SIP device？
A1:
1) Register SIP soft phone
Download the x-lite softphone from counterpath website
www.counterpath.com
After install the x-lite, right click the panel and sel ect the SIP Account setting and then configure it.
Display Name: 500 User Name: 500 Password: 500 Authorization Name: 500 Domain: 192.168.5. 150
2) Register IP Phone (for example, Yealink’s T28 IP Phone)
a) Connect the T28’s WAN port to the switch. And it can get the IP from your route. b) Press the ‘OK’ key on T28 to get the IP of T28. c) Put the IP on web browser then you can enter the T28 configure page through this IP. d) Put the SIP extensions info on the T28 IP phones.
Display Name: 501 User Name: 501 Register Name: 501 Password: 501 SIP Server: 192.168.5.150
Use the same method register another T28 to other extension.
Q2. How do I reset MyPBX back to the factory default settings?
A2: To perform a reset, please follow steps below: Step 1: Hold down the ‘Reset’ button on the back of the unit for 5 seconds
and watch the LEDs on the front of the MyPBX. When the status LED turns red, let go of the reset button. Step 2: When the RUN status LED starts blinking, MyPBX will be set back to factory defaults. Step 3: T o access the confi guration page, navigate to 192.168.5.150 using a web browser. Make sure that you are on the 192.168.5.0 subnet before doing this. Step 4: Login to th e device w ith the username ‘admin’ and the password ‘password’, in order to begin reconfiguring the device.
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APPENDIX B How to Configure Autobackup
Before Autobackup can be properly configured, an SMB share folder accessible from MyPBX must be set up on a Wi ndows based machine. Once that has been set up, please follow the steps below. Step 1 Add a new folder, rename it, and set this new folder’s share Properties accordi ng to Figure B-1
Figure B-1 Set up share Properties
Step 2 Enter the new folde r and create a new text file, then rename thi s file to status.txt. This step is very important, DO NOT forget to create the status.txt file.
Step 3 Configure Autobackup settings on MyPBX to Figure B-2
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Figure B-2 Autobackup Setting
SMB share host/IP: Change this to the IP add re ss of the computer where backup files will be stored. SMB share name : Change this to the name of the shared fold er where backups will be stored. SMB share us e rna me : The user name used to log into the network share. Leave this blank if it is not required SMB share password: The password used to log into the network share. Leave this blank if it is not required
If configuring Autobackup for the first time, the system will prompt you to restart MyPBX once your settings have been sa ved. After rebooting , open your Wi ndows sh are f ol de r to see if the MyPBX backup file s and f ol ders has been created. If the conte nts of the backup folder look si milar to Figure B-3, then you have successfully configured Autobackup on the MyPBX unit.
Figure B-3 Autobackup setting succeed
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APPENDIX C How to Configure NAT setting
When MyPBX is behind a NAT(firewall),you need to configure NAT setting on M yPBX if yo u want to use a remote extension. Please follow section 1 or 2 below depending on your network configur ation.
1. If MyPBX is connecte d to a l ocal network, you must set up port forwarding
on your router. Specifically, you must map port 5060 (default SI P port) and port 10001-10200 (default RTP port range) as UDP ports.
Next, go to the MyPBX web interface and configure the SIP settings according to Figure C-1:
External IP: yo ur router’s pu blic IP address External Host: y ou r router’s domai n External refresh: 20 seconds Local Network Address:192.168.5.0/255.255.255.0 (change this
according to your network configuration)
NAT mo de: Yes Allow RTP Reinvite: No
Figure C-1
Assuming that your router’s host address is yeastar.3322.org, your local network is from 192.168.5.1-192.168.5.254, and the subnet Mask is
255.255.255.0, the MyP BX network settings should configured like Figure
C-2
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Figure C-2 MyPBX Network setting
2. If MyPBX has a public IP, (i.e. is connected directly to your internet
service pr ovider), the network settings should be configured according to Figure C-3:
Figure C-3
Next, you should configure the NAT settings according to Figure C-4
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Figure C-4
External IP: The public IP address of MyPBX External Host: Leave this blank if no domain has been configured External refresh: Leave this blank
Local Ne twork Address: Leav e this blank
NAT mode: Yes Allow RTP Reinvite: No
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APPENDIX D How to Use Auto Provision
Step1. Disable DHCP Server on your local network. E.g. Disable DHCP Server on Linksys Router.
Figure D-1
Step2. Enable DHCP Server on MyPBX. Login MyPBX web interface, System Settings DHCP Server  Enable DHCP Server.
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Figure D-2
Step3. Configure phon es on MyPBX auto-provision page.
1. Login MyPBX web interf ace, Basic  Auto Provi sion  Create New Phone.
Figure D-3
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2. Fill in the phone detail message on the pop-up windows.
Input IP Phone’s MAC addre ss, configure Name, Call waiting, Line, Extension, Label, Line active for the phone.
Figure D-4
Step4. Turn on the power and connect the network cable to IP Phone. Remark: The factory default sett ing of DHCP for IP Phone is enable, so you can
skip this step to step 5. If the DHCP is disable, please follow below step to enable it. (e.g.: Yealink’s IP Phone).
1. Login IP phone’s web page.
2. Enable DHCP.
Figure D-5
Step5. Finish.
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APPENDIX E How do I configure distinctive ring tones
Step1: On your IP phone, navigate to the Phone settings web configuration page and find the Distinctive Ring Tone sectio n. For each cust om ring t one, ent er the Int ernal Ringer Text (can be digits or text) to trigger the ring tone. For example, you may enter “Family”. e.g.: Yealink’s IP phone.
Figure E-1
Step2. Configure the ‘Custom Ring ’ on MyPBX. MyPBX web interface, Inbound Route s Edit Inbound Route, f ill in the Int ernal Ringer Text on ‘Custom Ring’.
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Figure E-2
Step3. Finish.
<Finish>
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