Руководство по настройке (базовым функциям), Englis

AP-IPNext SMT Manual

AddPac Technology OSTS

2010 AddPac, All Right reserved. AddPac Technology and AddPac logo are trademarks of AddPac Technology intellectual property.

Contents

1. Introduction of IP-Next Series

2. Initial Settings

3.

User’s Registration

4. Setup for Supplementary Service (IPNext)

5. Setup for Supplementary Service (External Servers)

6. Appendix

A. IPNext HDD setup (Fdisk / format / Raid)

B. IPNext Initialize

C. IPNext Redundant Configuration

2 AddPac Technology

Front View

1)

2)

3)

POWER LED

LAN 0,1 LED

FXO LED

IPNext Exterior (Front View)

1) POWER LED, 3) FXO LED

2) LAN0,1 LED

1) FastEthernet(Wan), 2) Console. 3) FXO 0

Parts of Rear Side

1) Power Switch

1) Power, 2) FastEthernet(Lan) 3) USB 4) FXO 1

3

Rear Side

AddPac Technology

Major Features of IPNext

Major Features of IP-Next

Features

Telephony and

Service

&

Features

Contents

• Voice Mail

• Call Parking

• Call Pickup

• Call Forwarding

• Auto Attendant

• Calling Number and Name Identification

• Call Transfer – Blind, Consult by Softkey

• Call Waiting Indication Call Swapping by Softkey

• Call Hold by Softkey

• Conference Control (internal/external MCU)

AddPac Technology Proprietary &

Documentation


Features

Contents

Advanced

Features with AddPac IP phone,

Video Phone, etc

• with AddPac IP phone, Video Phone, etc

• Plug and Play with Auto Discovery Function (SSCP)

• Softkey Map Download and Control (SSCP)

• Time and Date Setting

IP-PBX

Signaling

Protocols

• SIP Application Server, Proxy, Registrar and Location Server(RFC3261)

• Multiple ITSP Trunk with SIP & H.323 Account Support

- IP UA Client Role for Registering to ITSP SIP Server

- H.323 Gatekeeper Client Role for Registering to ITSP H.323

Gatekeeper Server


Documentation


Features

IVR(Interactiv e Voice

Response)

&

Auto

Attendant

Voice Mail

Conference

Contents

• Default Auto Attendant Support

• IVR Function

- Provides with GUI-based Smart IVR Scenario Editor

- Upload/Download Scenario by Smart IVR Scenario Editor

- Supports Multiple Concurrent Scenarios

- Support Recordable IVR Prompts

• Support Voice Mail with IVR

• Access from Remote Site via Trunk Support

• Voice Mail Notification Support

• Ad-Hoc Conference

• Dial-Out Conference

• Meet-me Conference

• Multiple External MCU Support (Video, Audio, etc) : AP-MC1000


Documentation


Features

Music &

Announcement

• Music on Hold

• Replaceable Announcements

• Dialing Music/Tone Service

Contents

IP-PBX User

&

Device

Management

• Auto Discovery of IP Phones & IP Video Phones

• Monitoring Status of Phones

IP-PBX

Miscellaneous

Function

• Auto Config & Upgrade

• Personal Directory (Smart Messenger)

Basic Routing

• IPv4/IPv6 Dual Stack

• Management Routing

- Telnet, FTP, TFTP, SNMP, Syslog support

- Packet filtering (Access-list)

- Static Routing


Documentation


Features

Contents

Network

Management

• Standard SNMP Agent (MIB v2) Support

• Traffic Queuing

• Remote Management using Console, Telnet.

Security

Functions

• Standard & Extended IP Access List

• Enable/Disable for Specific Protocols

• Auto-disconnect for Telnet/Console Sessions


Documentation


Features

Operation

&

Management

Other

Scalability

Features

Contents

• System Performance Analysis for Process, CPU, Connection I/F

• Configuration Backup & Restore for APOS Managements

• System Booting and Auto-rebooting with Watchdog Feature

• System Managements with Data Logging

• IP Traffic Statistics with Accounting

• Debugging command

• DHCP Server/Client

• Network Address Translation (NAT) Function

• Port Address Translation (PAT) Function

• Cisco Style Command Line Interface(CLI)

• Network time Protocol(NTP) Support


Documentation

Hardware Features of IPNext

Details

CPU

Contents

• High Performance RISC Integrated Host Processor

Memory

LAN Interface

PSTN Interface

Power &

Operation

Environments

Main Memory

• 128Mbyte SDRAM

Flash Memory

• 512Mbyte

SDRAM Memory

• 512Kbyte Flash Memory

LAN0 Port

• One(1) Ethernet 10/100Mbps Fast Ethernet

LAN1 Port

•One(1) Ethernet 10/100Mbps Fast Ethernet

Console Port

•One(1) RS-232C Interface for CLI

FXO Port

•None (Model A)

•2-Port FXO Voice Interface (2 * RJ11)(Model B)

• Power Requirement: Power VAC 110~220 VAC, 50/60Hz, 5V 3A

• Operating Temperature: 0°C ~ 50°C (32° ~ 122°F)

• Storage Temperature: -40°C ~ +85°C (-40° ~ +185°F)

• Relative Humidity: 5% ~ 95% (Non-condensing)

Dimensions

• 38(H) * 182(W) * 182(D)

Weight

• 0.46 Kg


Documentation

Function of Signaling / Media Proxy

( RTP Proxy)

NAT router

Private

200 201 202

IPNext

Private

NAT router

300

301

302

PSTN


Documentation

Smart Multimedia Manager

Smart Multimedia Manager

addpac propriety

SMM message

SMM

Local Area

SMM

IPNext

100

101

102

103 200 201

Major Functions

202

• IP-Next Registration all setups related to user.

• IP-Next all setups related to service

• IP-Next status review

• IP-Next message trace

300

301

302

PSTN


Documentation


Smart Multimedia Manager

Category Sub-Category

Administration

Mode

Advanced mode

Easy mode

Event monitoring

Real Time Monitoring for the

Call being Processed

Real Time message Trace

Redundancy

Message logging

Active-Active

Active-Standby

Terminal

Registration

IP

SSCP

Generic SIP terminal

SIP Trunk

V6

V4

Description of Functions

Display Mode for setting up all the functions of IP-PBX

Mode for Setting up only Basic Functions of IP PBX

Monitoring Function for Enabled Calls at the present time

Real Time Message Tracing Function for Calls Being Processed at the Present

Time

Storing and restoring Functions for Calls being Processed (filtering function)

The structure allowing Two IP-

PBX’s to provision service simultaneously

The structure allowing one of two IP PBX’s become master and the other one become disbled

The terminal using AddPac Specific Terminal Control Protocol. AP-IP-XXX, AP-

VP-XXX series come under this subcategory

Supporting linking protocol of SSCP and general SIP terminal or SIP gateway with

Multiple Ports

Trunk Gateway or Service Provider’s SIP server for interoperability with PSTN network outside

IP Version 4

IP Version 6

AddPac Technology Proprietary & Documentation


Category

Codec Class

Device Profile

Sub-Category

codec list by group

Account

Device

Management

Administrator Account/password register/delete

Subscriber account/password register/ delete

Phone User user General

Terminal General

Trunk Gateway General routing pattern

Call Control

SIP Proxy Server General

Register (option)

Functional Description

Each codec class can be set to each group

In case of SSCP terminal,

Reference can be made for SSCP terminal after common options are grouped and a profile is created admin account which can access to SMM access account for Smart Messenger or voice mail assigning telephone numbers to subscribers

Detailed information on telephone subscriber (one’s real name, subscriber id, password and others)

Setup and binding of profile/partition/VoIP Protocol for telephone subscribers

Setup and binding of location/device profile/partition/VoIP Protocol for Trunk gateway

Routing pattern input(Digit map)

Call priority / authority / MRBT/CID… location/device profile/partition/RTP Proxy/Network domain/SIP

ID/password/DTMF relay method and other setups

Register Phone number



Category

MCU

Subcategory

Local MCU

Remote MCU

Auto Upgrade Auto Upgrade Server

Option

Call Policy pick-up partition

Partition access list

Number Translation Rule

Routing group

Routing List

Registering Local MCU

Functional Description

Registering Remote MCU

Assigning auto upgrade server outside or IP PBX

Auto upgrade group/URL/server position/ access information (id/password) and others.

Setup for enabling direct pickup in the same group only for pick-up group

Used for setting ranks for outgoing calls of each group/ subscriber

Used for setting ranks of outgoing calls of each group/ subscriber

Registering patter list of change in number

Routing Tag for Trunk interface

Routing pattern list to be applied to Routing Group


Initial Setting

1. Basic configuration(CLI)

2. LdapClient set up(CLI)


Console and Telnet Login

Telnet ethernet

Baud rate 9600

No parity

1 stop bit

No flow control

FastEthernet 0/0

COM port

Serial port console

Default login account/password : root/router

Command of change Root account password

IP-PBX> enable

IP-PBX# config t

IP-PBX(config)# no username root

IP-PBX(config)# username root password addpac administrator

% Caution: After no username is set and rebooted, the root account disappears and remote access becomes impossible. The only way to access is automatic login, the authority at this time is administrator

% Must save after setup.

IPNext


Installation guide for IPNext

•

Booting messages in IPNext(via console)

System Bootstrap, Version 1.2

Decompressing the image:

##########################[OK]

?System Boot Loader, Version 5.1.6

Copyright (c) by AddPac Technology Co., Ltd. Since 1999.

[DUAL-BOOT] Start application (0xbd000000)...

System Bootstrap, Version 1.2

Decompressing the image:

#################################################

AddPac IP-PBX Series (IPNext_G2)

32BIT RISC Processor With 125MHz Clock

128 Mbytes System Memory.

512 Kbytes System Boot Flash Memory

32 Mbytes System Flash Memory

1 RS232 Serial Console Interface

…

Press RETURN to get started.


Basic configuration

•

Connect IPNext

Welcome, APOS(tm) Kernel Version 8.xx.xxx.

Copyright (c) 1999-2008 AddPac Technology Co., Ltd.

Login: root

(Login ID)

Password:******

(Default login Password : router)

IP-PBX>

•

Set hostname

IP-PBX> enable

IP-PBX# configure terminal

IP-PBX(config)#

hostname <Name> (set Hostname )

IP-PBX(config)#

•

Set Server Clock

IP-PBX> enable


IP-PBX(config)#

clock time <Year><Month><Day><Hour><Minute><Second> (set clock)

IP-PBX(config)# clock time 2010 03 15 15 28 45

IP-PBX(config)#


Basic configuration

•

•

Set Interface IP

IP-PBX> enable


IP-PBX(config)#

interface FastEthernet 0/0

IP-PBX(config)#

ip address <IP Address> <Subnetmask> (Set ip address)

Example)

IP-PBX(config)# ip address 172.17.109.200 255.255.0.0

Set Gateway

IP-PBX> enable


IP-PBX(config)#

ip route <Destination Network IP> <Wildcard Mask><Gateway IP>

(set Gateway ‘s ip)

IP-PBX(config)#

Example)

IP-PBX(config)#

ip route 0.0.0.0 0.0.0.0 172.17.1.1


LdapClient set up(CLI)

•

Necessary configuration for running Smart Multimedia Manager(WEB)

IP-PBX#


IP-PBX(config)#ip tcp keep-alive count 5

IP-PBX(config)# ip tcp keep-alive idle 30

IP-PBX(config)# ip tcp keep-alive interval 5

IP-PBX(config)# no http authentication

IP-PBX(config)# http document-root /hd

IP-PBX(config)# ldapclient

IP-PBX(config-ldapclient)# name IPNext

IP-PBX(config-ldapclient)# host 127.0.0.1 389

IP-PBX(config-ldapclient)# ldap enable

IP-PBX(config-ldapclient)# end

IP-PBX#

This name will be display in SmartClient

Set host IP(local loopback) + 389 (port)

IP-PBX(config)# network-domain interface ip FastEthernet 0/0 domain public

IP-PBX(config)# network-domain interface ip FastEthernet 0/1 domain private

* FastEthernet

0/0 set Public IP with “Public "domain name

FastEthernet 0/1 set

Private IP with “Private” domain name for terminal which have Public IP and

Private.


User’s Registration

1) Creating Subscriber’s Extension Numbers

2) Creating Subscriber’s Extension Users

3) Opening of Subscriber’s Terminals

4) Registering Subscriber’s Terminal

5) Trunk Setup(including Service Provider’s signaling server )

6) Number Routing table


The Procedure for Registering Subscribers

Creating New Subscriber Numbers (Phone Number)

Registering to Subscriber Terminal (Terminal)

Assigning Subscriber Numbers to Subscriber Terminal

Creating Trunk (inbound/outbound SIP Server or TG)

Setting up for Number Routing Map

Confirming Registration and Testing Calls


SMM Login

The Main Page http://172.17.109.201

via explorer

“Click” Icon Click

ID : root

Password : router

It can change server port using Setting button

If you changed port, need to change IPNext also (via CLI)

24

SMM Main Page


Smart Multimedia Manager Operation

Creating New Subscriber Numbers (Overview of Creating New Extension Number Bands)

1.

2.

IPNext

Series can create a highly sophisticated and flexible numeric system which are suitable to any client’s organizational structure and operational policy.

Let’s create the extension numbers for an imaginary client’s company organizational structure which can be

described below:

Each division will have 3 different extension numbers from X001 to X103

Each division of employees use VPare accommodated by IPNext

300’s which

B u s i n e s s A d m i n i s t r a t i o n

( e x t e n s i o n 1 0 0 0 ~ 1 0 0 3 )

3.

The operator allocates the numbers created for

VP-

300’s which are accommodated by IPNext

R & D ( 2 0 0 0 ~ 2 0 0 3 )

3. Creating the new extension numbers

Local Area

AP-VP300N

S a l e s ( 3 0 0 0 ~ 3 0 0 3 )

2. Requesting extension numbers

SMM

(Smart Multimedia Manager)

1. The new extension numbers are created

P r o d u c t i o n ( 5 0 0 0 ~ 5 0 0 3 )

Technical Support(6000~6003)

AP-IPNext


SMM – Call Manager(Phone Number)

•

Select Add Phone number menu

1. Add Phone Number

3. Input Start number and End number

2. Select Range Option

26

4. Push “OK” button for create numbers


SMM – User Management(Phone Users)

•

User Group Create

- Create user Group for management.


SMM – User Management(Phone Users)

1. Create User information

•

Mandatory field

– Last Name

– User ID

– User Password

– Voice Mail Password

2. Assign phone number for each User

– Select the number for user using find button

3. Set Nick name which will use in Smart Messenger


Smart Multimedia Manager Operation

1.

2.

3.

A basic testing for call connection can be performed as to allocate the created number sources previously to 5units of AP-VP300N

Each IP phone can be set up by each separate IP and temporary setup files can be used for the basic setup

Please refer to the next page for more details of setting up IP Phone.

Local Area

Requesting

registration

Requesting

registration

Requesting

registration

Requesting

registration

Requesting

registration

Business Administration

X1000

R&D

X2000

Sales

X3000

Production

X5000

Technical Support

X6000


Documentation

AP-VP300N setting

•

Setup AP-VP300N

1) Go to network menu -> Internet select IPv4(DHCP, Static, PPPOE) or IPv6(Static, EUI-64, Auto Configuration)

2) Go to network menu -> SSCP

Input IPNext(Master

)’s IP-address in Call-manager 1 field change option of SSCP setup(off => on)

3) Go to network menu -> Presence -> Presence Server input

IPNext’s IP-address in Presence Server 1 field input “5051” in Server1 port

4) Go to network menu -> Presence -> Presence Setup input User ID, User password then set Presence Enable to “On”


AP-VP300N Button information

Creating New Subscriber Numbers (testing call connection between extension numbers-

Setting up AP-VP300N by using OSD)

Description for Overall Key Features of AP-VP300N

No. Keys

(1)

(2)

(3)

(4)

(5)

(6)

(7)

(8)

(9)

(10)

(11)

(12)

(13)

(14)

(15)

(16)

(17)

SpeedDial

F1~F4

INFO

Number

Key

Hook Flash

Camera

Privacy

Menu

Cancel

Call

방향키, OK

END

View

Absence

Voice Mail

HDP Call

SPK Call

Features

Brining a menu for speed dial

When you use Phone Book or Speed Dial, these functions are assigned to the touch screen, which is located in the lower part of the screen.

While on the line, you can select whether to display information

Dialing by using remote control. Used for entering each value of setup of

OSD menu

Hold key for headset or speaker phone during phone conversation

Used for turning on/off the camera screen

While on the line, the other person is not able to see you

Brining OSD main menu

Used for moving to a higher category in OSD menu.

Used for brining call history from the latest and calling by pressing number keys

Checking and moving with in OSD menus or adjusting camera movement

To end on line call

Changing conversation view/ remote view/ local view while on the line

Used during the user’s absence (to be released)

To listen to voice mail (to be released)

For using headset

To use imbedded speaker phone


Documentation

AP-VP300N OSD setting


Setting up AP-VP3 00 by using OSD)

1) Entering <Main Menu> by pressing menu key

2) Moving to <Network> by using direction key.

3) Moving to <Internet> by using direction key(left, right) Pressing “OK” key to enter <Internet> sub-menu


Documentation




1) Moving to <SSCP Setup> by using direction key

2) Pressing OK key to enter <SSCP Setup> sub-menu


Documentation




1. Pressing ‘1’ to change to ‘release’ from <SSCP setup>

2. After moving to <Call-manager1>, IP addresses of Call-Manager (IP-PBX) can be entered.

3. After returning to <SSCP setup>, ‘setup’ can be changed by pressing ‘1’

4. Completing <SSCO Setup> by pressing ‘OK’ key


Documentation



Setting up AP-VP3 00 by using CLI) configure terminal

osd

sscp enable

sscp

call-manager 1 10.1.1.1 5060


Documentation

SMM - Device Management

1.

2.

You can request to register those terminals, which has been set up, to IPNext.

You can make reference of this registration by entering ‘Device Management > Terminals’ which is shown below:



1.

2.

3.

If you doubleclick ‘Terminal Name’ you can see the window of ‘Terminal Properties’ opens up as it is shown below:

From ‘General’ tab, the basic setup can be made for naming the terminal and automatic registration.

After the setup is made in the screen below, you may doubleclick ‘Phone Number’ for assigning the numbers.



1.

The designated port of the terminal can be selected, then the number sources can be referenced by a doubleclick.

Double Click



1.

Selecting the number to be assigned to a designated terminal as it is shown in the picture below:

Supporting list for recently

subscribed numbers

Number Assignment

39

The number to be assigned to a designated terminal



1.

2.

Confirming the assigned number as it is shown in the picture below

Through the procedure of confirmation (authorization granted) by the operator, the designated number can be set automatically to IP Phone



1.

After finishing assignment to the designated terminal as it is shown in the picture below, it completes the process of registering the terminal.


SMM - Trunk

Trunk Gateway is an equipment to take the call, in case the inside user makes an outgoing call. IPNext can be used as an inbound Call Manager, making an outgoing call to outside by other equipment (VoIP, PSTN) is possible after registration to Trunk Gateway. For making an outgoing call or taking an incoming call, the setup of Trunk Gateway is mandatory. Routing Pattern needs to be set up for fixing the telephone number to be dialed for the user to be connected with the external terminal. In general, when you want to exit to outside, you may press the number 9 and this is one of the cases. The picture below shows the screen and registration information for Trunk Gateway to work.


SMM - Trunk

⑫

⑬

⑭

SMM-Device Management-General Registration to Trunk Gateway

Ref. Description

1

Registering a name of Trunk Gateway .

2

①

Registering a description of Trunk Gateway

3

Selecting

‘Device Pool’ (Basically set to default)

②

4

Selecting

‘Location’ (basically no setup is provided)

③

④

5

6

Selecting ‘Network Domain’ for Call Manager

Selecting IP version

7

Registering ‘IP Address’

⑤

⑥

⑦

8

9

Selecting VoIP protocol

Entering a designated port for VoIP protocol

10

⑧

⑨

⑩

Setting up a type of

‘DTMF-Relay’ to be transmitted to Trunk

-Transmitting DTMF based on Rtp-2833 : RFC-2833 standards

-In-band: Transmitting DTMF tone through RTP

-Transmitting DTMF by using SIP INFO, H.245 Signal

⑪

11 Keep Alive Method Option

-Disable : don’t check Trunk Gateway status

-Register : check Trunk

Gateway’s status using SIP REGISTER messages from Trunk Gateway

-Option : check Trunk

Gateway’s status using SIP REGISTER messages from Trunk Gateway

12

This is a setup to provide MOG for the terminal requesting hold

13

This is a set to provide MOH for interoperability with Nortel

’s soft switch

14

This is a setup by force, for using RTP Proxy while the telephone is on the line


SMM - Trunk

⑩

⑦

⑧

⑨

①

②

③

④

⑤

SMM-Device Management-Trunk Gateway- Registering Information of Call Control

⑥

6

7

8

9

Ref.

1

Description

Selecting Partition Access List for setting up authorization of inbound call.

2 Selecting Partition Access List for setting up AAR(Automatic Alternate Routing)

3 Setting up priority for inbound call

4

5

10

Selecting Inbound Access Rule for setting up authorization of inbound call rejection using calling number pattern

Applying Translation Rule for outgoing and incoming call numbers to change number for inbound call.

Selecting an option of provisioning MRBT(Ring Back Tone) for inbound call

Selecting an option of presenting a number of outgoing call to outbound call

Default: Default setting of Call Manager (by following Call Manager Cluster >

Options > Calling Party Presentation)

Allowed: Presenting my number to the other

Restricted : Not displaying my number to the other

Setting up a number of outgoing call by force (ie.: Representative Number)

Setting for don’t generate CDR information, when you use more than 2 IP-PBX as

Trunk, don’t generate outbound call for each Trunk(IP-PBX) (if you don’t set this option, the user’s can be charged both of IP-PBX )

Setting up an option for whether Trunk is to interoperate with outbound


SMM - Trunk

SMM-Device Management-Trunk Gateway- Registering Information of Failed Call Redirection

Ref.

1

Description

Set for Failed Call Redirection (SIP 302 Moved Temporary) function

①

②

③

2

3

Set IP-address for IP-PBX get happen Failed Call Redirection Messages to transfer

Set

Server’s port information when IP-PBX send Redirection call

④

4 Set redirection function using specific reason


SMM – Trunk(example)

•

General Tab a.

Input Device Name b.

Select Network Domain (Public / Private)

-

It depend on T/G’s IP address. c.

Set IP-address d.

Select Signaling Protocol(SIP / H.323)

•

Call Control Tab a.

Inbound Call

I.

Select Number Translation(Incoming call) b.

Outbound Call

I.

Calling party presentation or Caller ID DN


SMM - Trunk

This is a function to register SIP Proxy Server for interoperating with the other SIP Proxy Server in outbound.

For example, when an outgoing call is made by registering a company telephone number to SIP Proxy Server, this function can be set up to enable the call which has been registered to SIP Proxy Server.


SMM - Trunk

①

②

⑦

⑧

⑨

⑪

③

④

⑤

⑥

⑫

⑬

⑭

⑮ ⓐ ⓑ

SMM-Device Management-SIP

Proxy General Registration

⑩

Ref.

1

2

3

4

5

Description

Registering a name of SIP Proxy Server

Registering a description of SIP Proxy Server

Selecting

‘Device Pool’ (Basically set to default)

Selecting

‘Location’

Registering SIP User Name

6

7

8

9

Registering SIP password

Registering

‘Local Domain’.

Selecting ‘Network Domain’.

Setting up a type of

‘DTMF-Relay’ to be transmitted to Trunk

-Transmitting DTMF based on Rtp-2833 : RFC-2833 standards

-In-band: Transmitting DTMF tone through RTP

-Transmitting DTMF by using SIP INFO, H.245 Signal.

10

IP address and protocol list of SIP Proxy Server can be set up and priority level can placed

11

The setup value of Register Expire time to be used for registering SIP Proxy Server

12

This is a setup by force, for RTP Proxy to be used while telephone is on line.

13

Using local host name instead of domain name of Proxy Server for registration:

To : 7000@local hostname From : 7000@local hostname

14

Registering username instead of the number e164 for the information to be registered to

Proxy Server

To : [email protected] From : [email protected] Contact : [email protected]

15

This is a setup to provide MOH for hold request from a terminal

A

B

This is a set up to provide MOH for interoperability with Nortel soft switch

This a set up whether to

‘Register’ or not for the Phone Number to be registered to

Proxy Server


SMM - Trunk

SMM-Device Management-Registering Information of SIP Proxy Phone Number

②

1


The list of telephone numbers which are not registered to SIP Proxy

Server

2 The list of telephone numbers to be registered to SIP Proxy Server

User Name:

Registering user name for the telephone number to be authorized

Password: Registering password for the telephone number to be authenticated

Status: status of registering the telephone number

①


SMM - Trunk

SMM-Device Management-Registering SIP Proxy Call Control

⑨

⑦

⑧

④

①

②

③

⑤

⑥

1


Selecting Partition Access List for authority setup for inbound call

2

Setting up priority level for the inbound call

3

4

5

Selecting Inbound Access Rule for setting up authorization of inbound call rejection using calling number pattern

Applying Translation Rule to outgoing and incoming call numbers to change the number for the inbound call.

Selecting an option of presenting a number of outgoing call to outbound call

Default: Default setting of Call Manager (by following Call Manager

Cluster > Options > Calling Party Presentation)

Allowed: Presenting my number to the other

Restricted : Not displaying my number to the other

6

7

8

9

Setting up a number of outgoing call by force (ie.: Representative

Number)

Setting up whether to use P-Asserted-Identity header or not

Setting up whether to use P-Asserted-Identity in SIP From header field

Setting up whether Trunk is to interoperate with outbound or not


SMM - Trunk

SMM-Device Management-Registering SIP Proxy Failed Call Redirection

①

④

②

③

2

3

4

1

Re

f.

Description

Set for Failed Call Redirection (SIP 302 Moved Temporary) function

Set IP-address for IP-PBX get happen Failed Call Redirection Messages to transfer

Set

Server’s port information when IP-PBX send Redirection call

Set redirection function using specific reason


SMM - Routing Pattern

•

Add Routing Pattern

– Input routing pattern

• Lind line

• Mobile

• Toll

• International

• Etc

– Select Trunk

– Select number translation rule

•

Provide outside Dial Tone

– If client want to use out-side dial-tone which provide(Sip-server or G/K) need to check.

•

Block this Pattern

– Set this routing pattern for blocking

– Users can’t use this routing pattern


SMM - Number Translation Rule

•

Create Number Translation rule

Add Translation rule a.

Input name b.

Add translation rule

I.

Input Matched Pattern (a condition)

II.

Substituted Pattern(a result)

Ex)

Input Matched Pattern [2-9]T

Substituted Pattern 02%01%99

User input output

2345 -> 022345

1234 -> 1234

T = no limitation(All number)

. = single digit (any number)

[] = range


SMM - Number Translation Rule

•

Create Translation Pattern a.

Input Translation pattern b.

Select number translation (can select translation rule before you made it)

I.

Called Number

II.

Calling Number

•

Translation rule can set both of

Called Number / Calling Number

• it can edit Translation rule using

“

Edit

”


SMM – Test call

Make Test call

Video-Phone(AP-VP300N) with Video

Conference Terminals(AP-VC5000) ,

Smart Communicator with Video-Phone

AP-VP300N with PSTN, etc.

Local Area

PSTN

AP-VC5000 AP-VC5000 AP-SMP1000 AP-VP300N AP-VP300N AP2120N


SMM - Setups for Additional Call

Services(IPNext)

•

Additional Call Services

– Hunt Group

– Pickup Group

– Park

– Park Group

– Music Announcement

– Auto Attendant

– Service Code

– Day Template

– Monitoring


SMM - Hunt Group

Hunt Group

– Priority Level - Overview

3.

4.

5.

Hunt Group is a function to take a call, basing on a level of priority placed on members of this group, when the call is made to a specific and representative number. While some of the members’ lines can be busy or during their absence, the call can be delivered according to the following level of priority:

1.

2.

A call is sent from 1001 to 2000.

The number 2000, as the number of Hunt Group 2000, it consists of each member of the numbers, 2001, 2002, 2003.

Among these number, the call is delivered first to the number with the highest level of priority which is the number

2001

When the number 2001 is busy or absent, the call can be delivered to the 2002

When the number 2002 is busy or absent, the call is to be delivered to 2003

When the number 2003 is still busy or absent, the call can not be delivered and is to be terminated.


SMM - Hunt Group

•

Set Hunt Group

1.

Carrying out ‘Call Management>Supplementary Service>Hunt Group>Add Hunt Group’ of SMM Menu

2.

Creating an inbound group number of Tech_Hunt (Hunt Group Number 1099 )

3.

Setting ‘Hunt Mode’ to ‘Preference’, ‘Simultaneous’, ‘Random’.

4.

Selecting the extension number of the subject group by choosing a tab of ‘Phone Number’.

5.

Checking whether the call is processed in the order of the setup by dialing from 1001 to 1003.


SMM – Pickup Group

Overview of Pickup Group

This is an enhanced function from ‘Call Pickup’ of the same group, which serves to pick up a call for the other group.

When a call came to the other group, you can pick up a by pressing ‘Pickup’ button.

The number for ‘Pickup’ must be created differently from the user number

1.

2.

3.

A call is made from 1001 to 2001.

Call Manager takes the message, then send it to 2001

When the bell rings on 2001, you can press ‘Group Pickup’ from 3002 (2001 and 3002 are different members of

‘Pickup Group to each other’.

4.

1001 is connected to 3002.

Pickup

Group 1

IP 300N

1001

(1) Call 2001

(4) Confersation

IP 300N

1002

(2) Call 2001

(3) Group

Pickup

IP 300N

2001

Pickup

Group 2

IP 300N

2002

59

IPNext 50

IP 300N

3001

IP 300N

3001

Pickup

Group 3


SMM - Pickup Group

1.

Carrying out Call Management>Supplementary Service>Pickup Group>Add Pickup Group of SMM Menu.

2.

Creating a group extension number of Technical Support division as ‘Pickup Group Number 2999’

3.

Selecting an extension number of the subject group from the tab of ‘Phone Number’.

4.

Completing the setup, then dialing from 1000 to 6000

5.

Checking the response on 1000 by calling 6999 from an extension number 2000


SMM - Park Address Pool

Call Park-Overview of Park Address Pool

This function is useful in a situation of which you may need to move to one place to another during phone conversation.

During phone conversation, you may press ‘Call Park’ key, the ‘Park’ number is notified, then the line is off the line.

Then you can take the call from the place to where you move and the call is sent to the ‘Park’ number, your last call is

connected.

1.

2.

3.

4.

1001 and 2001 are on the line.

2001 pushes ‘Park’ key (1001 listens to a sound on hold while waiting)

Call Manager informs the ‘Park’ number.

After moving to 1002, you can press the ‘Park’ number

5.

The call is connected to 1001.

(1)Conversation

(2) MOH

(2) Parking

IP 300N

1001

IP 300N

2001

(3) Parking

Adress 8000

(5) Group Park

(4) Parked Calll

IPNext 50

IP 300N

1001

IP 300N

2002


SMM – Park Address Pool

Call Park-Park Address Pool-Setup

1.

Selecting ‘Call Management>Supplementary Service>Park Address Pool’.

2.

Selecting Add Park and setting up ‘Park Number’.

3.

Making a call from1000 to 6000.

4.

6000 reposes to the call then pressing ‘Park Number’, then pressing ‘Park Number’ from the other seat and connecting to 1000.


SMM - Park Group

Call Park-Overview of Park Address Pool

This function is useful in a situation of which you may need to move to one place to another during phone conversation.

During phone conversation, you may press ‘Call Park’ key, the ‘Park’ number is notified, then the line is off the line.

Then you can take the call from the place to where you move and the call is sent to the ‘Park’ number, your last call is

3.

4.

5.

connected.

1.

2.

1001 and 2001 are on the line

2001 pushes ‘Group Park’ key (1001 listens to a sound on hold while waiting)

Call Manager informs to all and each terminals included in the group

After moving to a seat, you can choose ‘Group Park’, then pick up.

The line connected to 1001.

IP 300N

1001

(5) Group Park

(3) MOH

(1)Conversation

(2) Group Parking

(4) Parked Calll

IP 300N

2001

(4) Parked Calll

IPNext 50

(4) Parked Calll

IP 300N

1001

IP 300N

2002


SMM - Park Group

Call Park

– Park Group -Setup

1.

‘Call Management>Supplementary Service>Park Group’

2.

Setting ‘Park Group Name’ by choosing Add Park Group

3.

Selecting the phone to be registered to ‘Park Group’.


SMM – Music & Announcement

This is a setup to enable/disable MoH (Music on Hold), messages of voice guide and dial tone policy.

The following service is shown in the picture below as to be carried out (Call Management > Advanced

Service > Music & Announcement Service) .

①

②

③

④

Ref.

1

2

3

Description

Enabling or disabling Music & Announcement Service. When the service is connected, Dial Tone can be furnished by Codec from

IPNext200.

Selecting Codec which can be suitable to VoIP environment

(G.711a, G.711u, G.729)

Setting a number of frames for each packet

⑤

4 Setting a policy of dial tone which can be applied to SSCP supporting IP Phone.

-Local Dial tone: Furnishing dial tone from IP phone during Hook off

-Remote Dial tone: Furnishing dial tone from Call Manager during

Hook off

-Local + Remote Dial tone: Dial Tone can furnished form IP Phone during Hook Off and replacing the dial tone by the one which is furnished from Call Manager

5

-Enabling or disabling Announcement when call fail.


SMM - Auto Attendant

As a process which is used in a general office environment, this is a service which allows you to listen to introduction of company by using IVR service, to make a call by entering a designated extension number and when you do not know the extension, you may press ‘0’ to be connected to information desk.

A simple type of IVR service, which is provided at default, is presented in here and for more complex type of IVR service can be used by IVR Editor which is used for editing the scenario.

The following service can be carried out (Call Management > Advanced Service > Auto Attendant

Service) , as it is shown in the picture below.

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②

③

④

⑤

⑥

3

4

1

2


Enabling or disabling Auto Attendant Service

Setting up the telephone numbers for Auto Attendant Service

Selecting ‘Partition’ (options)

Selecting the audio codec which is to be used in Auto Attendant Service.

The supporting codec is G.711A/Ilaw and G.729

5

6

Setting a number of frames for each port

Setting up a designated extension number for when ‘0’ is entered from

Auto Attendant Service


SMM – Voice Mail

This service configures the settings for allowing the registered use, to the IPNext, to listen to the message at any time and any where, when the user is away from the phone or busy on the line.

The service can be performed by locating Call Management > Configuration > Advanced Service > Unified Messaging

Service as it is shown in the following figure.


①

②

③

1

2

3

Enable or disable the Unified Messaging Service

Configure the phone numbers and partition for connecting to the message box.

Configure the phone numbers and partition for connecting to the user’s message box without authenticating the extension number.

4

Configure the leaving phone number and partition for the Unified Messaging service.

④

5

Choose the option for audio codec to be used for the Unified Messaging service

G.711A/Ulaw and G.729 are the supporting codec.

⑤

⑥

⑦

6

7

Choose the option for video codec to be used for the Unified Messaging services

MPEG4, H.263, H.263+ and H.264 are the supporting codec.

Set the number of frames for each Packet


SMM - Service Code

Service Code Configuration can assign 2 digit numbers at maximum starting from ‘#’ or ‘*’, sets up the supplementary service numbers, which are supported from IPNext. This configuration enables/ disables the supplementary service and more useful to the terminals not supported by SSCP, than the IP phone using Softkey which is supported by SSCP

⑦

⑧

⑨

⑩

⑪

⑫

⑬

⑭

⑮ ⓐ ⓑ ⓒ ⓓ ⓔ ⓕ

①

②

③

④

⑤

⑥

7

8

9

10

11

4

5

6

1

2

3

Ref.

A

B

C

D

12

13

14

15

E

F

Description

Set up the code for

Set up the code for Call Pickup

Set up a call to the code which does not belong to the same group for Call Pickup

Set up the code for the phone user’s absence

Set up the code for rejecting the incoming call

Deactivating the code from Call Reject (Absence and Do not Distub)

Set up the code to activate Call Waiting

Set up the code to deactivate Call Waiting

Set up the code to register the forwarding number unconditionally

Set up the code to register the forwarding number for the busy line

Set up the code to register the forwarding number during the phone user’s absence

Cancel the code for Call Forwarding

Set up the code to activate Call Forwarding unconditionally

Set up the code to deactivate forwarding unconditionally

Set up the code for Call Forwarding when the line is busy

Set up the code to deactivate Call Forwarding when the busy line

Set up the code to activate Call Forwarding when the phone user is absent

Set up the code to deactivate Call Forwarding for absence

Set up the code to deactivate Call Forwarding for absence

Set up CCBS

Cancel CCBS


SMM - Monitoring

•

Monitoring

–

Call Service Monitoring

–

Active Call Monitoring

–

Call Trace Monitoring

Call Service Monitoring

Call trace Monitoring



Call History searches the history of CDR log file which is saved in Smart Event Manager, Call Manager or PC.

With the information of phone number and host, this feature finds the call flow which has been processed at a glance.

To search the data, specify the event source (Menu > Tools > Preferences > Event) and operate.

To save the syslog to the SMM, use Save Event Log.

CDR enable

Result of Call history



SMM provides and reporting functions and the statistical data of Call Usage, Number of Calls, Incompletion Call and

Rank, basing of the basic information that IP-PBX provides.

Call usage Statistics

InCompletion Call Statistics

Number of Call Statistics

Ranking Statistics


SMM Setups for Additional Call

Services(Additional Servers)

1) Conference Management

2) Presence Management

3) IVR Management


SMM - Conference Management

MCU(Multipoint Conferencing Unit) Module is device connecting conference calls.

The type of conference call can be categorized as voice and video conference calls depending on a type of media.

IPNext is equipped with a module for voice conference call and an additional device is required for video conferencing call.

There are MCU-specific product such as the AddPac MC3000 and MCU imbedded in terminal product such as

VC2000.



⑤

⑥

⑦

⑧

⑨

⑩

Settings of Ad-Hoc Defaults are already registered basically when Conference Rooms is performed. Then select, double-click and perform Properties from the pop-up menu This Conference Room is the configuration used for Ad-Hoc

Conference from the system, so it can not be deleted.


③

①

②

⑪

1

2

3

4

Register a name of Conference.

Register a description of Conference

Select a type of media (Audio, Video + Audio) for Ad-Hoc Default. According to the type of media is selected, the settings of video can beenabled or disabled.

Select Media Class to be applied to Conference Room. Media Class can be set to 3 participants at maximum and 1 must be selected at least. Media Class takes a priority as the fist Media Class does not support, the second one does.

④

5

6

Set a priority for the Conference

Set a range of participants who can take the floor.

-All Participants can speak

-All Visible Participants : all the participants displayed on the layout can speak

-Floor + Chair : Only the participants with the floor and Chair can speak

Floor Only : Only the participants with the floor can speak

7

8

9

10

11

Set encryption mode (AES, DES, 3-DES) for video and voice data of

Conference

Allow the terminals which do not support encryption to participate into

Conference

This is an option to select whether the conference call is to be terminated when Chair

Man, who invites the Conference Call, terminates a call when he/she ends. When the option is selected, Conference Call ends when Chair Man terminated his/her call.

When the floor does not exist in Conference, Set the screen for the speaking participant to be stretched automatically

This is to configure the settings of video for the media type which is Video + Audio.

-Terminal : These are the characteristics of video transmitted from the participating

Terminals (optional)

Layout : Configuration of the layout for video



⑧

⑨

⑩

⑪

⑫

①

②

③

⑤

⑥

⑦

Configuring Dial-Out Conference

⑬

1

2

3

4

5

Ref.

6

7

8

9

10

11

12

13

Description

Register a name of Conference.


Enter a number of Conference

Select a type of conference. In here, select Dial-Out Conference.

Select a type of media (Audio, Video + Audio) for Ad-Hoc Default. According to the type of media is selected, the settings of video can be enabled or disabled.

Select Partition (optional)

Select Media Class to be applied to Conference Room. Media Class can be set to 3 participants at maximum and 1 must be selected at least. Media Class takes a priority as the fist Media Class does not support, the second one does.

Set a priority for the Conference

Set a range of participants who can take the floor.





Set encryption mode (AES, DES, 3-DES) for video and voice data of

Conference

This is an option to select whether the conference call is to be terminated when

Chair Man, who invites the Conference Call, terminates a call when he/she ends. When the option is selected, Conference Call ends when Chair Man terminated his/her call.


This is to configure the settings of video for the media type which is Video +

Audio.

-Terminal : These are the characteristics of video transmitted from the participating Terminals (optional)



⑧

⑨

⑩

⑪

⑫

⑬

⑭

①

②

③

④

⑤

⑥

⑦


•

Configuring Meet Me Conference

⑮

Ref.

1

2

3

4

5

6

7

8

9

10

11

12

13

14

15

Description

Register a name of Conference


Register a conference number.

Select a type of conference. In here select Meet-Me Conference.

Select a type of media (Audio, Video + Audio) for Ad-Hoc Default. According to the type of media is selected, the settings of video can be enabled or disabled.

Select Partition (optional), Select Media Class for Conference Room

Select Media Class to be applied to Conference Room. Media Class can be set to 3 participants at maximum and 1 must be selected at least. Media

Class takes a priority as the fist Media Class does not support, the second one does.

Set the maximum number of participants (1~16 participants)

Set the priority level for Conference

Set the range of the participants taking the floor

-All Participants : All the participants can take the floor

-All Visible Participants : All the participants who are displayed in the layout can take the floor

-Floor + Chair : Only the participant with Floor and Chair can take the floor

-Floor Only : Only the participant with Floor can take the floor

Set the encryption mode (AES, DES, 3-DES) for video and voice data of the

Conference

Choose the option for terminating the conference call, when Chair Man, who invites the Conference Call, terminates a call when he/she ends. When the option is selected, Conference Call ends when Chair Man terminated his/her call.


Select an option whether to open or close the room. When the room is closed, set a password. (4 numeric digits)

Configure the settings of video for the media type which is Video + Audio.


-Layout : Configuration of the layout for video


①

②

③

④

⑤

⑥

⑦

⑧

⑨

⑩

⑪

⑫


Configuring Ad-Hoc Dial-Out Conference

⑬

Ref.

1

2

3

4

5

6

7

8

9

10

11

12

13

Description

Enter a name of the Conference

Enter a description of the Conference

Enter the Conference Number

Select the type of conference. In here, select Ad-Hoc Dial-Out Conference

Select the type of media (Audio, Video + Audio) for Ad-Hoc Default.

According to the type of media is selected, the settings of video can be enabled or disabled.

Choose the option for Partition (optional), Select Media Class for Conference

Room

Choose the option for Media Class to be applied to Conference Room. Media

Class can be set to 3 participants at maximum and 1 must be selected at least. Media Class takes a priority as the fist Media Class does not support, the second one does.

Set the priority level for the Conference

Set the range of participants who can take the floor.





Set encryption mode (AES, DES, 3-DES) for video and voice data of the

Conference

This is an option to select whether the conference call is to be terminated when Chair Man, who invites the Conference Call, terminates a call when he/she ends. When the option is selected, Conference Call ends when Chair

Man terminated his/her call.


Configure the settings of video for the media type which is Video + Audio.





Make Conference room

– Select Conference number(don’t use same phone number)

– Select Media type (Audio only, Audio + Video)

– Enable Voice Switching

• If you enable this function. The speaker video will display larger than others

– Close on Chair Out

• The Chair disconnect the conference, the room will be destroy.



•

Video

– Terminal

• This function can select Terminals bandwidth, frame rate, pic size.

– Layout

• Set layout of this conference.



•

Select participant for this conference room

•

Set one number as a Chair(when chair disconnect this conference this conference will be destroy)


SMM - Presence Management

Speed Button assigns a telephone number to a specified button and automatically makes a call to the phone number.

It is referred as the button information in both hardware and software aspects.

AP-VP300N support this speed button. This feature creates a profile and registers information to each button and the button information can be provided to the subscribers’ terminals through Presence Server.

Speed Button Profile can be created in Presence Group.

Ref.

1

Description

Enter a name for Speed Button Profile

①

②

③

2

3

4

Enter a description for Speed Button Profile

Change the position of the speed button to higher or lower in the order

Enter a name to be displayed for each speed button(1~25)

④ ⑤ ⑥ ⑦

5 Choose the type type for each speed button (1~25) of phone number

6 Enter a telephone number for each speed button (1~25)

(Pressing the right click, you can search and enter the telephone numbers which have been registered)

7

Use the features of Soft Key(Redial, NewCall, Hold, Transfer)on the speed button. This options is possible if Speed Button Type is set to

Soft Key


SMM - Presence Management

Presence group is a unit to provide presence service and is able to group the settings of speed button and a range of information sharing between the users. One user’s directory can configure many presence groups, share the speed button profiles and take presence message. The same profiles are applied for redundancy.

Create Presence Group

1.

General

- Input Presence Group Name

2.

Boundary

- Choose Organization or User Group

3.

Speed Button

- Add Speed Button profile

①

②

82

③


SMM – Unified Messaging Management

Unified messaging server provides the feature allowing the registered user to the IPNext PBX, to listen to the left message at any time and any where, when the user is away from the phone or busy on the line.

With accordance to message type, the property is categorized by voice and video and the server provides the features

of message box management (resetting, creating, moving and deleting) and message management

(moving and deleting).

As it is shown in the following figure, the list of unified messaging servers are displayed by choosing the menu

(Unified Messaging Management > System > Unified Messaging Server):


①

②

③

④

⑥

⑦


Message Box Configuration

Message box configuration is the overall settings related to message box and message box for leaving the voice or video message.

(Unified Messaging Management > Configuration > Message Box Configuration)

⑤

Ref.

1

2

3

4

5

6

7

Description

Set a length of time (seconds) for saving voice message

Set a length of time (seconds) for saving video message

Specify the maximum HDD capacity of the message box for the phone number

Specify a type of message to be left for the phone number (Audio or

Audio + Video)

* The present version does not support audio and video, but it will supported in future

Set the quota for exceeding the maximum capacity of HDD of the message box. The old message can be deleted or the new message can be disregarded.

Set Email Notification when the message is left. The recorded file can be attached to the e-mail. The server file can be deleted after notification

Choose the option for blocking the message box for the failed password entry. A number of password entries can be specified. If the message box is blocked, the message can not be left or verified.



Unified message box references, searches or manages all the message boxes created on address basis.

In message box list, the assigned users or a number of messages left , HDD usage can be verified and the box provides management (initialize, move and delete) for each address of the mail boxes.



Unassigned address searches and references the address list without the message box created, and creates the message boxes collectively.


Create Selected

Message Boxes

Create the message boxes for the selected address collectively

Refresh

Refresh the unassigned address list


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②

③

④

⑤

⑥

⑦

⑧

⑨

IVR Management > Configuration > IVR Service By pressing the right button of the mouse, your can register a new IVR service.

⑩

6

7

8

9

10

2

3

4

5

1


Create a name of the IVR service

Enter a description for IVR Service

Create a number of the IVR service

Choose the option for a name of IVR Server Cluster.

Choose the option for a name of IVR scenario.

After the scenario is created by Smart IVR Editor, then IPNext 700 is configured and only the registered scenario list can be selected,.

Choose the option for partition (optional)

Choose the type of audio codec

Set the number of frame for each packet

Choose the option for enabling the IVR service

Go to IVR Schedule mode


The following figure displays the screen for the IVR schedule list registered to the IVR Service Number.

The IVR scenario is serviced at a specific date and time.

A schedule template can be registered for the specific date and time When they are configured, they can be serviced basing on the schedule template. The corresponding details are disabled, because there is not need to specify a particular date and time for the schedule.

The day template can be also registered (Call Management > Advanced Service > Day Template) for a specific date.

After the day template is configured, the IVR scenario is serviced at the registered date in the template.


1

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①

2

3

4

5

6

7

8

The registered schedule list. Many schedules can be registered for each

IVR Service Number

Configure the registered schedule template

.Configure the starting date and time of the registered schedule

Configure the registered day template

Configure the days of the registered schedule

Configure the starting and ending time of the registered schedule

The basic IVR scenario of the corresponding Service Number

Set the priority level for the service of the registered schedule

⑦


Example of Smart IVR Editor

The main screen is displayed after executing Smart IVR Editor

1) Creating a NewProject

shows the New Project window for the setting of each property. Click ‘OK’, then the scenario ID with the same directory under the specified path is created and well as the basic Project within.



2)

Add ‘Get Number menu’

You can also create the diagram as to drag and drop from the Diagram Tool Box.



3 ) Connecting Diagram

Each Diagram needs to be connected to display the order and branch conditions. All the Diagrams can have more than one connection. Depending on the types of diagram, many braches can be connected. Connecting each

Diagram can be processed in the following procedure:



4) Set Get Number Menu

The user takes a specified number of DTMF input. Beside the input, the Diagram can be moved for the specified

input like Menu Diagram, once the Event Digit is set. Each branch does not need to be connected directly and it can be connected automatically by the properties settings, like Menu Diagram.


1

2

3

8

9

10

4

5

6

7

11

Name Displays a name of Get Number Diagram

Ment File Register the Voice File to be heard to the user before the one take the input. To cancel the registered Voice File, place the cursor on the icon, the press the right click of the mouse to can the Ment File.

Cancelable Cancel the Voice File which has been set up at ② When

Cancelable is selected and the user makes an entry, the Voice File is stopped playing.

Play Plays the registered Voice File at ② .

Digit Count Set the number of DTMF which the user can enter

End Digit Set up DTMF which informs that the entry is completed. Enter

End Digit DTMF for the number less than the one which has been set at ⑤.

Event Digit Set up the DTMF to move to another Diagram without taking the entry.

When the Event Digit DTMF is entered, it moves to the Diagram for the

Event Digit of the Event Condition.

Initial Timeout Set the timeout for the user’s first entry. Display ‘No input

Event’ if no entry is made within the time range.

Inter Digit Timeout Set up the Timeout between the user’s entries. Display

‘No match Event’, if no entry is made in the range.

Allowable Count Set the allowable count for no match and no input in the

Event Condition. When the count is less than the count of no input, no match, then request the user for more inputs. When the count is more than the one, the move to the specified Diagram

Event Condition Set the branch condition for occurrence of the Event by the user’s input. OK is for receiving the input normally, the Event Digit is for receiving the Event Digit value. For more details of the branch condition, please refer to the branch condition of the Menu Diagram



5) Add ‘Processing’ and two ‘play’ menu with make link



5) Add more menus which connect with high level















7) Build the Project

When you are finished with writing out the scenario by using the Diagrams and Links, you can perform Project Build.

You may click Project -> Build (or Ctrl + B) to start Project Build.

When the Project Build is finished, a message of success is displayed as it is shown

8) Set Connection Configuration

Set server information

- Server IP(IPNext)

- FTP information (User ID: root / Password : router)

- Http information (User ID: root / Password : router)

- check Use Authorization



7) Project Upload to IPNext

When the Project Transfer window opens as it is shown in below, set the Upload from the Transfer type and then click OK. You may see the window opens up for 3 types of Codec format to be selected. when uploading the Project is completed.


Appendix

• IPNext (Initialize Ldap (SMT))


Initialize Ldap (SMT)

• Ldap install (SMT)

2. Set the IPNext’s IP address to system host field

Don’t need to change Port and user ID

( if you change IPNext’s http port it need to

set same port )

1. Main page for maintenance tool

Click to “Next”


Initialize Ldap (SMT)

1. Display IPNext 700’s information

2. Choose System recovery

3. Choose System initialize

4. Select Full initialize

System Maintenance tools Cont.



2. If you already using IPPBX function, all IPPBX function be stop

1. Display which function will do.


3. Stop all IPPBX function with restore all data to IPPBX



2. Show all System initialize completed

”Click OK”

1. After System initialize it need to reboot

“Click OK”

3. Display System initialize completed

“ Click Exit”



IPNext Stand alone Test

2. PC will download SMM

automatically when click icon.

1. Access IPNext via web-browser http://172.17.109.200:80

3. after download it’ll display login page

login ID : root

password : router


After login Web SMM

IPNext Stand alone Test

Stand alone Simple Test Scenario.

1.Add 2~3 user datas..

2.Add 2~3 tel numbers.

3.Register 2~3 terminals.

4.Make a call each other.

5.Make backup file.(excel or ldap)

6.Add more information.


Thank you


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