Chapter 5: System Feature

Voicemail

Details configuration on Voicemail: Voicemail Reference/ Voice Message Options/ Playback Options. If you need to send message by mail to your defined mailbox, you must configure SMTP andEmail model.

Click Voicemail to display the dialog as shown below

Item	Explanation
Max Greeting Time(sec)	Maximum Greeting Time
Dial “0” for Operator	Dial “0” to cancel the voicemail and forward to Operator.
Message Format	Save the voice message as this format, WAV (16-bit) or Raw GSM.
Maximum Messages	Maximum messages to be allowed to leave.
Max Message Time(min)	Maximum Time for each message to be allowed to leave.
Min Message Time(sec)	Minimum Time for each message. The message will be deleted automaticallyif the time is less than the minimum message time.
Say Message Caller ID	Checking this option, Caller ID will be played when user login email toreceive the voice message.
Say Message Duration	Checking this option, the message duration will be played before playing thevoice message.
Play Envelop	Envelop includes date, time and caller ID.
Allow Users to Review	Check this option to allow users to review the voice message.

Voicemail to Email Settings

Item	Explanation
Attach voicemail to Email	The voicemail will be sent as attachment to the user’s Email.
Sender Name	The sender’s name will be displayed when you receive the Email.
From	Mailbox to send email
Subject	Subject of the Email.
Message	Input the Email template.

Conference Bridge

A conference bridge allows a group of people to participate in phone call. The most common form of bridge allows participants dial into a virtual meeting room from their own phone. Meeting rooms can hold dozens or even hundreds of participants.

This M-200 supports 3 conference rooms. Please configure it on this page

Item	Explanation
Conference Number	The number that users call in order to access the conference room; the defaultnumber is “900”.
Conference Password	Password for users to access the conference, e.g.”1234″.
Administrator Password	Password for administrator to access the conference.
Conference DialPlan	Use this dial plan to invite other participants.
Play hold music for the first participant	Check this option to play the hold music for the first participant in theconference until another participant enters this conference.
Enable caller menu	Check this option to allow the participant to access the Conference Bridgemenu by pressing “*” on the dialpad.
Announce callers	Check this option to announce to all Bridge participants that a new participant is joining the conference.
Record conference	Recorded conference format is WAV.
Quiet Mode	If this option is checked, all the participants in the conference can hear only, but it is not allowed to speak.
Leader Wait	Wait until the conference leader (administrator) enters the conference before starting the conference.

Please check the following diagram to learn:

Go to conference:

In the conference, admin can add new participant (extension user or external number) to the conference.

In the conference, the administrator can invite new guest (extension user or external number) to the conference. (Default password for admin is 2345)

Learn how to invite new guest to the conference as the diagram is shown below:

Paging and Intercom

Paging and Intercom is used for calling a paging extension; all terminals which support this function will be picked up automatically and listen; meanwhile, it supports duplex.

Click Advanced -> Paging and Intercom -> New Paging Group

Item	Explanation
Paging Extension	The number users will dial to page this group.
Description	Provide a descriptive title for this Page Group.
Paging Group Members	Selected device(s) on this page
Device List	Select Device(s) to page.
Duplex	Paging is typically one way for announcements only. Checking this will makethe paging duplex, allowing all phones in the paging group to be able to talkand be heard by all. This makes it like an “instant conference”.

NOTE:

For Paging/Intercom function extension (IP phone), support Auto Answer

DISA

A trunk call is made to the PBX, and call is made to another trunk through outbound route of the PBX. This trunk can make international calls. You are out of the office and want to contact your customerin a foreign country. Now you can dial DISA number after PIN authentication. You are now connected to your customer, and you can speak to your customer now.

Click DISA -> New DISA to display the dialog as shown below

Item	Explanation
Name	Define a name for DISA.
PIN Set	User will be prompted to input this number when PIN Authentication isneeded.
Record in CDR	Check to record.
Response Timeout(sec)	The maximum time for waiting before hanging up if the dialed number isincomplete or invalid. Default is 10 seconds
Digit Timeout(sec)	The maximum interval time between digits when typing extension numberis 5 seconds by default.
Extension for this DISA(Optional)	If you want to access DISA by dialing an extension, you can define anextension number for this DISA.
Select Dial Plan	Select the Dial Plan for this DISA.

Phone book

Manage your Phone book and set the Phone book Contact for speed Dial.

Click Advanced -> Phone book

Create Contact

Using Phone book for Speed Dial

E.g. prefix is *99, speed number is 00, destination telephone number is 2345678. When dialing *9900,

The call is going to 2345678 automatically.

Callback

When user makes calls by the callback number to M-200 IPPBX, the call will be hung up automatically. Then the PBX will call back this number and forwarded to define destination after the callis connected. Please configure it as shown below

Click Advanced -> Callback

At first, enable this function. Select Dial Plan, and define the callback rule (strip digits or prepend prefix). Click New Callback Number to add callback number

Input callback number and define the destination

Smart DID

Smart DID: After extension user makes an outbound call, the call is ringing back to IP PBX, and directed to the one who made the last call. Please configure it as shown below

Click Advanced -> Smart DID