Miracall Technology MC-IP8008 User Manual

Miracall Technologies

IP PBX MC-IP8008
User Manual

Please read this manual carefully before operation

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TABLE OF CONTENTS

Chapter1 Brief Introduction & Features ........................................................................ 2

1.1. System Features ................................................................................................. 2

1.2. Specifications ...................................................................................................... 3

1.3. Default configuration ........................................................................................... 4

Chapter2 Basic Configuration........................................................................................ 5

2.1. Log on to the system........................................................................................... 5

2.2. Configure User Extensions ................................................................................. 6

2.3. Trunk ................................................................................................................... 7

2.4. Outbound Routes............................................................................................... 10

2.5. Inbound Routes ...................................................................................................1 1

2.6. IVR (Interactive Voice Response) ..................................................................... 12

2.7. Record .............................................................................................................. 13

2.8. Voicemail........................................................................................................... 14

2.9. Conferencing..................................................................................................... 15

2.10. Music On Hold .................................................................................................. 16

2.11. Call Parking....................................................................................................... 16

2.12. Ring Groups ...................................................................................................... 17

2.13. Time Based Rules ............................................................................................. 18

Chapter3 Status Display ............................................................................................... 18

3.1. Call Logs ........................................................................................................... 19

3.2. Register Status.................................................................................................. 19

3.3. System Info ....................................................................................................... 20

Chapter4 System Management .................................................................................... 20

4.1. Network and Country ........................................................................................ 20

4.2. DDNS Setting............................................................................………….......... 21

4.2. Change Password............................................................................................. 21

4.3. Backup .............................................................................................................. 22

4.4. Upgrade ............................................................................................................ 22

Chapter5 Operating Instruction ................................................................................... 23

5.1. How to make a internal call ............................................................................... 23

5.2. How to make a outbound call ............................................................................ 24

Make call via PSTN trunk................................................................................ 24

Make call via VoIP trunk .................................................................................. 26

5.3. How to make a incoming call ............................................................................ 27

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Chapter 1 Brief introduction & Function

MiraCALL04/08 is an embedded ippbx based on standard asterisk for SMEs, which
is not only a PBX, but also as a voice mail Server, IVR server, conferencing server. With 4
or 8 analog interface which can be configured as FXS or FXO ports (made in factory),
and 1Wan and 1Lan with router function. With excellent echo canceller function, it can
meet most of the customers requirement.

1.1 System Features

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Based on Asterisk

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Configuration by Web

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Built-in SIP/IAX Server

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Static/DHCP/PPPoE network access

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Codec: G.711-Ulaw, G.711-Alaw, G.726, G.729, GSM, SPEEX

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SIP/IAX Trunk(use with VoIP Trunk operator)

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Zap Trunk(Use with PSTN)

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SIP/IAX Extensions(connect with IP Phone)

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Zap Extensions(connect with Analog Phone)

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Voice Mail ,and voicemail to email

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Flexible Dial Plan

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Call Conference

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Auto-attendant

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Music On Hold

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Call Queue

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Call history list

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Support IP Phone with Key function

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FAX T.38

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Voicemail to email

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Other basic function:

1.Three way conference

2.Call Forward(and forward to external PSTN or mobile)

3.Call Hold

4.Call Transfer

5.Call Waiting

6.Caller ID

1.2 Specifications

1) Interface

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4or 8 * Analog Ports (RJ11)

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1 * USB Interface(for Wifi moudule, unavailable)

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1 * SD MMC Interface (2GB,to install voicemail)

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2 * Network Interface (RJ45,1 for WAN, 1 for LAN)

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1 * Power port（DC 12V 2A）

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1 * Re-set Button (beside PWR button,black)

2) Panel

Mark Function Status Description

On Power On

PWR

Power Status

Off Power Off
On System working

SYS

System Status

Off System disabled
Wink Data exchanging

WAN

WAN interface Status

Off No Data exchanging
Wink Data exchanging

LAN

LAN Interface Status

Off No Data exchanging
On MMC enabled

MMC

SD card Status

Off MMC disabled
On WiFi enabled

USB

USB WiFi module

Off WiFi disabled
Red FXO port
Green FXS port

1-8lamp

Analog Modules Status

Off FXO or FXS no connection

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3) Hardware

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32bit embedded RISC DSP

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256M Onboard Nand Flash

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2M Onboard Nor Flash

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64M Onboard SDRAM

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2G MMC/SD Storage

4) environmental requirements:

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temperature: -10 °C -48 °C

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Storage temperature: -30 °C -68 °C

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humidity: 10-80% no dew

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Power: 90~260VAC

5) Packing List

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MC-IP4008,8008 1 Unit

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Power Adapter 1 Unit

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CD 1 pc

6) Packing Size:

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Inner box MEAS: 280*198*70mm

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G.W/unit: 1.3kg

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Carton MEAS: 520*580*220mm

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G.W: 15kg

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PCS/CTN: 10PCS

1.3 Default configuration

1. Wan port IP address: 192.168.1.100:9999

2. Lan port IP address: 192.168.10.100:9999

3. Web GUI username: admin

4. Web GUI password: admin

5. Command line username: root

6. Command line password: myippbx

Chapter2 Basic Configuration

2.1. Log into the system

After connecting the ippbx to the local area network. Launch the web browser on a

computer that is in this local area network. Enter the IP address for the system (Wan port
IP address 192.168.1.100:9999, Lan port IP address 192.168.10.100:9999) . The start
web page will appear like this:

Enter Username and password (default username is admin, password is admin), then

click login. Once the login is success the home page will be display:

With the MiraCALL GUI, you can configure extensions, conference, voicemail, Dial

Plan

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The home page is used for login out, Reboot or return to Factory default status. The home page is used for login out, Reboot or return to Factory default status.

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Logout: To log out the MiraCALL GUI.

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Logout: To log out the MiraCALL GUI.

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Restart Asterisk Restarted the Asterisk system.

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Restart Asterisk Restarted the Asterisk system.

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Reboot: Reboot the IP-PBX system

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Reboot: Reboot the IP-PBX system

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Factory Defaults: Restore all settings to de factory default.

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Factory Defaults: Restore all settings to de factory default.

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Activate change: Made the change active for the current configuration after you

make a configuration change on some page.Everytime you should activate after

change the command or setting.

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Activate change: Made the change active for the current configuration after you

make a configuration change on some page.Everytime you should activate after

change the command or setting.

2.2. Configure User Extensions 2.2. Configure User Extensions

Click the Users tab and you will see the extensions setting, your created user in this

page.

Click the Users tab and you will see the extensions setting, your created user in this

page.

The following information comprises a user extension definition:

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Extension The extension number given to station user.

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Name The name for this extension, can be only.

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Password Password to register the extension, same as IP phone

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VM Password Password used to access specified voicemail extensi on .

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E-mail Address Voice mails received by this extension can be sent as audio

file attachments emailed to a specific address (optional)

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Caller ID The Caller ID number displayed for intercom call

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Analog Phone To assign the analog phone to connect with which FXS port

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Dial Plan The dial restriction, conditions set for making calls

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There are also several advanced extension options available. The advanced options

establish the connections from the listed extension to other systems within the IPPBX
system server. These advanced options include the following:

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Voicemail The extension support voicemail or not.

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Can reinvite Can send call again In SIP protocol

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SIP The extension support SIP protocol

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IAX The extension support IAX protocol

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NAT Address reflection between the WAN and public network. When

extension is within internal network, it should use NAT

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Call Waiting The extension support Call Waiting function

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3-Way Calling The extension support 3-Way Calling functions

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Codecs Click here, you can set the extension’s codec (voice decode)

(default support: alaw, ulaw,g792).

2.3. Trunks

If you want to make external call, you must register with a Trunk in order to connect to

the Public Switched Telephone Network (PSTN) or other VoIP service provider. Through
the web page you can add a trunk.

When you click Add a Trunk , There are two Trunk categories: Analog
Trunk, Custom Trunk.

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Analog Trunk You can enable analog PSTN trunk by this option (FXO port).

You need to delete the default Ports 1~6, and Add a PSTN Trunk based on how many
FXO port you have actually.

The above map shows the IP PBX has only 2ports Analog trunk. After choose Port 1 and 2,
you need to click “Save”

, then to enable the setting.

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Custom VoIP Trunk The Custom VoIP option allows you to create a custom VoIP

definition. To create the custom VoIP provider definition you will

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need to complete the following:

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Comment The name of the SIP Trunk

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Protocol You can select a SIP or IAX protocol

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Register Enable/Disable register.

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Host The IP address provided by your IP carrier provider

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Username The user name provided by your IP carrier provider

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Password The password provided by your IP carrier provider

Once you have added a VoIP Trunk it will appear on the list of Trunk on the Trunk
page. There is an Options drop-down list associated with each Trunk listing. The Options
drop-down list allows you to edit or delete the Trunk definition, as well as further refine the
definition by choosing several advance options. Select either Codecs or Advanced to
further refine the definition.

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Codecs Codecs provide the ability for your voice to be converted to a digital

signal and transmitted across the internet.

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Advanced The following advanced options are available to further refine your

trunk.

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Trunkname Specify a trunk name if you want to refer to the service provider

definition as something other than specified in Comment

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Insecure This option specifies how connections to a service provider (host)

should be handled. Valid options are very/yes/no/invite/port. (Default is
very)

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Port The register request is sent through the port. (Default is

SIP:5060,IAX:4569)

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Caller ID The caller ID will be set to the value specified in this field

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From domain Sets default from: domain in SIP messages when acting as a SIP client.

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From user Sets default from: user in SIP messages when acting as a SIP client

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Contact Specifies a primary extension for call routing

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2.4. Outgoing Call (Dial Rules)

The Dial Rules tab on the left toolbar allows you to use basic pattern matching to

differentiate outgoing calls and route them accordingly (create different DialPlan).

Click on Add a Dial Rule to define a new DialPlan. The following dialog will be displayed.

A DialPlan is comprised of the following items:

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Rule Name Set a rule name (can be any)

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Place this call through Select a Trunk through which the call should be made

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Analog fallback Select a ba c k u p Analog trunk when 1st trunk failed.

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Dialing Rules The Dialing Rule gives you the ability to use basic pattern

matching to differentiate calls and route them accordingly. For instance, if a number
begins with 9256, and is followed by 7 or more digits, that would define a call within
the state of Alabama. If a call began with 9 followed by 7 digits, it would be a local call

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that probably didn’t require a long distance charge. Instead of adding a rule for every
extension or phone number you call, specify the pattern in this rule similar to the
example.

that probably didn’t require a long distance charge. Instead of adding a rule for every
extension or phone number you call, specify the pattern in this rule similar to the
example.

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Strip This option gives you the opportunity to remove specified

digits from the call being dialed and replace them with the digits needed to make the
call. (same like Dial 9 before making any outside call, like you dial 9+8008108000, in fact,
system will strip “9” and the actual number you dial is 8009108000). You can also
prepend digits to the beginning of this number (automatic add a digit, such as add 0
before national distance call). Each time after setting ,you should click save

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Strip This option gives you the opportunity to remove specified

digits from the call being dialed and replace them with the digits needed to make the
call. (same like Dial 9 before making any outside call, like you dial 9+8008108000, in fact,
system will strip “9” and the actual number you dial is 8009108000). You can also
prepend digits to the beginning of this number (automatic add a digit, such as add 0
before national distance call). Each time after setting ,you should click save

and Activate .

2.5. Incoming Calls (Incoming Rules)

There are only a few options you need to configure

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Route

All Unmatched incoming calls: any call
Incoming calls that match: the incoming numbers meet some rules,
such as some spam numbers, numbers starting from 135xxx ect

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From Provider

From PSTN port or SIP trunk etc.

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To Extension

8001- -User 8001

Means this incoming call will route to extension 8001

8002- -User 8002

Means this incoming call will route to extension 8002 (can be fax)

8028- -User 8028

Means this incoming call will route to extension 8028

8030- -User 8030

Means this incoming call will route to extension 8030

600- voicemail

Means this incoming call will route to voicemail recording.

900--Conference

Means this incoming call will route to conference group

Workingtime- IVR

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This incoming call route to OGM of working time

Time-rule -- Time

Means this incoming call will route to time setting rules.

Closed Time- IVR

This incoming call route to OGM of closed time This incoming call route to OGM of closed time

2.6. Auto-attendant Setting (IVR)

Through the web page, you can create Interactive Voice Response (IVR). IVR are

designed to allow for more efficient routing of calls from incoming callers.

Voice menus are constructed depending on your needs. Just like your business you

need to create the solution best suited to your customers.

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Name Set a IVR (OGM) name

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Extension Set an extension which can use this IVR (OGM)

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Please Select Select a IVR message, including the custom message.

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Dial other Extensions? Enable or dis-able to

dial other extensions.

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Keypress’ Event - Dial any single Key on the phone, then the call will route to…

Dial Selectable Routes Routes destination

0 Go to Extension 8001~8032
1 Go to Menu OGM IVR of Working time, Close time
2 Go to Ring Group 1~ x (you can custom Ringing Groups)
3 Custom You can define the routes as you like
4 Hangup Goes on-hook or terminate the conversation
5 Play Invalid Indicate invalid
6 Disabled No any action

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Optional
# Optional
t Optional

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Record OGM

2.7.

In the event that one wants to record custom menu prompts for the IP PBX, which

can be used in a IVR, the Record may be used.

A list of previously recorded menus is displayed. Here, the user may modify several
options

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Record Again

To record a new voice message by phones

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Play

To play the message in an extension phone

Delete

Delete the recorded voice message

closed.gsm

The welcome message to announce company in closed time.

welcome.gms

The welcome message to announce company in working time.

new.gsm

The same as welcome.gms

There are two options under “Record a new voice”

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File Name To give a name of the message.

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Extension Used for Recording To select use which extension to record the message.

2.8. Voicemail

Voicemail is an option available for every extension in the IP PBX system. The
relationship between the extension and the voice mail is established in the User Extension
section of the GUI. You can configure the voicemail through this page.

The Voicemail Configuration page displays all the extensions to the left, including
voice mail. Standard configuration information is also present, allowing you to confirm the
extension used to check messages as well as general parameters such as the following:

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Extension for Checking Messages Dial this number to access personal extension

Voicemail box. Need Name and Password.

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Before voicemail, greeting for The welcome message before recording

voicemail . The incoming caller will hear
this self-recorded message before leave
their message.

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Attach greetings for email Email the voicemail message to the

address you set

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Dial “0” for Operator Incoming callers press “0” to cancel leaving

message in voicemail and switch call to the
operator extension.

There are several options that can be specified to define the voicemail message in the
system.

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Including WAV (GSM), WAV(16-bits),Raw GSM

Message Format

The format of the voicemail which saved in the IP-PBX

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Set the max. messages to leave for each extension user.

Maximum Messages

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The maximum duration of each message .

Maximum Message Time

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The minimum duration of each message .

Minimum Message Time

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It will report the Caller’s number before message played

Say Message Caller-ID

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Say Message Duration

It will report the voice mail duration before message played

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Report the Date, Time, and Caller’s number before the

Play Envelope

voice mail message played.

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This option provides incoming callers to review their
message before saved and can be played back by the
owner of the voice mail extension. Standard options are
presented to you, allowing you to discard the message or
re-record it if you aren’t happy with it.

Allow Users to Review

2.9. Conference Call

Allow more extension people to make a conference call meeting.

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Action Explain Default

Room Extension Dial this extension to entry to conference group. 900
Pin Code Input this password before login in conference 1234
Admin Pin Code Administer can use this code to manage the conference

room.

1234

Conference Optional

Play hold music for
first participant

The first joiner hear the waiting music

Enable participant Menu The setting menu for conference parties extension
Announce Callers Make announcement for the joiners
Record Conference Record the conference meeting .
Mute Mode No voice mode
Wait for Marked VIP user The important user is going to join the conference
Set Marked VIP user To set the important user .

2.10. Music On Hold

Music: In default setting,there are music 1~10 .

Upload Music File : to upload a holding music from TFTP Server.

2.11. Call Parking

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Call park allow you to place the incoming caller on hold. Any extension can talk to this
parked caller. The parked party will hear the music.

Extension to Dial for Parking Calls: Set a Call Parking number. To park the call, the

incoming Caller should be transfer to this Parked
number.

What extensions to park call on: Set the number dialed for call park (eg: 701-720) ,

each time for Park, system will assign a number
automatically for the Parked party, dial this
number to retrieve the conversations.

Number of seconds a call can be parked for: Set duration the Parked party can be

parked for. It will automatically return to
the last conversation extension when
the parked time finished.

Pickup Extension: Set the code to pick up ringing extension.

Call transfer when no-ansewr: Set the duration over time on no-answer.

2.12. Ring Groups

The extensions can be a group . Each extension will ring within the same group for the incoming
call. Each ringing extension can pick up this call.

Define Ring Groups to Dial more than one extension

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Name Set a Ring Group name, can not input special characters.

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Strategy There is a drop-down list, you can choose Ring all or Ring in order.

Ring all: ringing to all extensions until call answered . Ring in order: ring
one by one

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Ring Group Members Add Ring Group member from available right side.

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When there’s no answer ,you can choose to route the call to

Voicemail, IVR or hand-up.

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Number to dial to access ringing group The access code to ringing group.

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Ring time for each/all extension The duration ringing for each extension in

the group, after duration finished ,
call will ring to another circle or
another extension.

2.13. Call Forward

This rules allow the extension forward to intercom or external line/number, Immediately or On Busy,
No-one Answer the call

2.14. Time Rules

To set the times and date of the system. You can select the route destination when the time
is matched or not.

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Chapter 3 IP-PBX Status

3.1. Call List

To show calling history, starting time, duration, answered or un-answered
status .

3.2. Register Status

If there’s IP address displayed then it mean the extension is registered successfully.

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3.3. System Info

To display the IP-PBX system data and information , version, Asterisk version etc.

Chapter4 System Management

4.1. Network and Country

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On this page you can set WAN, LAN interface information and country.

IP Assign: you can select STAT IC, DHCP and PPPoE three mode
NTP: Set NTP server address.
Country: Set your Country, and use the Country Zone

4.2. DDNS

On this page, you can set DDNS reference.

Notice: Now, it only support 3322.org server. More other servers, you can customize your
requirement

4.3. Change Password

On this page, you can change the administrator password (Default password: admin)

4.4. Backup

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On this page, clicking the “Take a Backup” button, you can backup nonce configuration

4.5. Upgrade

In this page you can upgrade system package and music file Upgrade System Package

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Enter The Package Name Set system package name

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TFTP Server IP address Set TFTP server IP

Upgrade Music File

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*

Enter The Music File Name Set you want upgrade music file name

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TFTP Server IP address Set the TFTP server IP

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Select Music directory Select directory that you want saved music file.

Chapter 5 Operating Instruction

5.1. How to make a internal call

Add a New User

Users-> New:

Set new user Extension is 8099….
Name, Password and Caller ID, etc……
Select Dial Plan is DialPlan1

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Use a SIP/IAX IP Phone registered with the user.
Then you can use 801 call 802 successful.

5.2. How to make a outbound call

To make an outbound call, we need to add a trunk first. There are two types of Trunk:

Analog Ports: FXO ports of MC-IP8008, connect to local PSTN
VoIP T runk: SIP or IAX trunk, connect to remote SIP/IAX server

In the MC-IP8008 I am using, the port1-4 are configured as FXO ports, port5-8 are
configured as FXS ports. When a port is configured as FXO port, the corresponding LED
shows RED. When a port is configured as FXS port, the corresponding LED shows

GREEN.

What are FXO and FXS?

FXS (Foreign eXchange Station) is an interface which drives a telephone or FAX
machine. FXS interfaces get phones plugged into them, delivery battery, and provide
ringing. FXS interfaces are signalled with FXO signalling. FXO (Foreign eXchange Office)
is an interface that connects to a phone line. They supply your PBX with access to the

public telephone network. FXO interfaces use FXS signalling. FXS interfaces are that allow
you to hook telephones to your PBX, and FXO interfaces allow you to connect your PBX to
real analog phone lines.

Make call via PSTN trunk

Add Analog Trunk

Trunk -> Add a Trunk:

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Add Dial Rule
In a Dial Rule -> add a Dial rule as below
Dial Rules

We have now added a Dial rule “Dial_PSTN” in the “DialPlan2”.
As we can see from the dialing rule of “Dial_PSTN”, all numbers start with 9 will be cut

the first digit (‘9’) .

After we have done above, in the extension we can dial 9 + local number to dial out via
SIP line.

Make call via VoIP trunk

Add VoIP service provider

Trunk -> Add a Trunk:
Add a Custom Trunk

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Add Dial Rule
In Dial Rules -> add a new calling rule as below
Dial Rules

Now we have added a new calling rule “Dail_Out_to_SIP” in the “DialPlan1”.
As we can see from the dialing rule of “Out_VoIP”, all numbers start with 0 will be cut the
first one digits (‘0’) and sent to my sip service provide voip.
The Out_VoIP is in the same DialPlan1. Since we have added this dial plan to the
extensions in above, we don’t need to add dial plan again.
So far when we have added two calling rules, any call start with 9 will be route to PSTN,
and call starts with 0 will be route to VoIP.

5.3. How to make an incoming call

Add an Incoming call. Click “add a incoming call rule”

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Select Route “All Unmatched incoming
calls” From provider “Port 1,2”

To extension “8018 – IP Phone” (here, you can select a extension, a IVR or others) Then,
if there is incoming call from Port1,2 channel the extension 8018 will ring

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