Welltech Computer Co., Ltd.
SIPPBX 6200
User Guide
Release 3.1.0
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CONTENTS
CHAPTER 1 SIPPBX 6200 SERIES INTRODUCTION............................................................ 6
SIPPBX 6200 APPEARANCE DESCRIPTION ............................................................................... 9
LOGON SIP PBX 6200........................................................................................................... 15
CHAPTER 3 SIPPBX 6200 SETUP WIZARD.........................................................................17
INITIAL SETUP WIZARD ........................................................................................................... 17
SYSTEM SETUP WIZARD ......................................................................................................... 21
HEADQUARTER/BRANCH OFFICE SETUP WIZARD..................................................................... 26
EXTENSION SETUP WIZARD .................................................................................................... 29
PSTN GATE WAY SETUP WIZARD ............................................................................................ 31
SIP TRUNK SETUP WIZARD .................................................................................................... 34
CALL PICKUP GROUP WIZARD................................................................................................. 36
OUTGOING CALLED NUMBER SCREENING WIZARD ................................................................... 37
CHAPTER 4 SYSTEM CORE REFERENCE.......................................................................... 40
SYSTEM ................................................................................................................................. 40
DEBUG .................................................................................................................................. 45
LICENSE ................................................................................................................................46
CDR ..................................................................................................................................... 46
AAA ........................................................................................................................................ 48
HTTPS CERTIFICATE ............................................................................................................. 49
EVENT NOTICE....................................................................................................................... 50
BACKUP/ RESTORE ................................................................................................................ 51
Backup Configuration (With License) .................................................................................... 51
Backup Configuration (Without License) ............................................................................... 52
Restore Configuration ............................................................................................................ 52
Compact Database ................................................................................................................ 52
CHAPTER 5 SERVICE REFERENCE.................................................................................... 53
CALL ROUTING....................................................................................................................... 53
DIGIT MANIPULATION .............................................................................................................. 56
CALL INTERCEPTION............................................................................................................... 58
DEVICE ACL.......................................................................................................................... 58
RTP RESOURCE GROUP ........................................................................................................ 59
NAT GROUP .......................................................................................................................... 60
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DNIS SCREENING GROUP...................................................................................................... 61
EMERGENCY CALL ................................................................................................................. 62
VOICE CODEC GROUP............................................................................................................ 63
APPLY CHANGE...................................................................................................................... 64
CHAPTER 6 ENHANCE SERVICE REFERENCE................................................................. 65
SYSTEM ................................................................................................................................. 65
INTERFACE............................................................................................................................. 66
SIP ....................................................................................................................................... 67
VOICE MAIL ........................................................................................................................... 70
AUTO ATTEND ........................................................................................................................ 73
CONFERENCE ........................................................................................................................ 73
ANNOUNCE SERVICE .............................................................................................................. 74
LANGUAGE............................................................................................................................. 75
WAKEUP CALL ....................................................................................................................... 76
FAX SETTING ........................................................................................................................ 77
DEVICE AUTO CONFIGURATION ............................................................................................... 78
OFFICE PROFILE .................................................................................................................... 79
Service Code.......................................................................................................................... 82
Pickup .................................................................................................................................... 85
Holiday ................................................................................................................................... 85
Work Time .............................................................................................................................. 85
Phone Book (Office shared phone book)............................................................................... 86
Dial Plan................................................................................................................................. 86
DEPARTMENT PROFILE ........................................................................................................... 87
EXTENSION............................................................................................................................ 87
GATE WAY............................................................................................................................... 96
UAC ................................................................................................................................... 101
BROADCAST GROUP............................................................................................................. 102
TOLL RESTRICTION............................................................................................................... 102
VOICE MAIL ACCESS ............................................................................................................ 105
FAX ACCESS ....................................................................................................................... 105
ANNOUNCEMENT.................................................................................................................. 107
VOICE FILE .......................................................................................................................... 108
CDR REPORT...................................................................................................................... 109
Call Detail Report................................................................................................................. 109
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Extension Summary..............................................................................................................110
Department Summary...........................................................................................................111
CHAPTER 7 CALL FLOW EDITOR......................................................................................112
CALL FLOW MENUS AND TOOLS .............................................................................................112
AUTO ATTENDANT & VMS FUNCTIONS ...................................................................................114
AA & VMS LEAVE MSG SAMPLE CALL FLOW AS FIGURE 7.2-15............................................. 125
AA & VMS LEAVE MSG SAMPLE CALL FLOW: ....................................................................... 128
VOICE MAIL RETRIEVE FUNCTIONS........................................................................................ 129
VOICE MAIL RETRIEVE SAMPLE CALL FLOW AS FIGURE 7.3-6 ................................................. 133
VOICE MAIL RETRIEVE SAMPLE CALL FLOW:.......................................................................... 135
SERVICE SETTING FUNCTIONS .............................................................................................. 136
SERVICE SETTING SAMPLE CALL FLOW AS FIGURE 7.4-4........................................................ 140
SERVICE SETTING SAMPLE CALL FLOW:................................................................................ 146
CHAPTER 8 SYSTEM CONTROL........................................................................................ 148
SYSTEM ............................................................................................................................... 148
SYSTEM TIME....................................................................................................................... 148
NETWORK............................................................................................................................ 150
Voice Gateway Setting:........................................................................................................ 152
SNMP................................................................................................................................. 154
DHCP................................................................................................................................. 154
ACCOUNT MANAGER ............................................................................................................ 155
PROVISIONAL IP................................................................................................................... 155
SERVICE .............................................................................................................................. 156
UPGRADE ............................................................................................................................ 156
LOGOUT .............................................................................................................................. 156
CHAPTER 9 SYSTEM MONITOR ........................................................................................ 157
SUBSCRIBER STATU S ........................................................................................................... 157
CALL STATIST ICS .................................................................................................................. 158
RTP STATUS ........................................................................................................................ 159
RTP STATISTI CS .................................................................................................................. 159
SERVER STATU S .................................................................................................................. 160
DHCP STATUS (USED FOR AUTO CONFIGURABLE DEVICE ONLY).............................................. 162
LINE OVERVIEW ................................................................................................................... 162
CONFERENCE ROOM ............................................................................................................ 163
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EVENT ................................................................................................................................. 164
DEBUG INFO ........................................................................................................................ 165
PING.................................................................................................................................... 166
CHAPTER 10 TELNET & RS-232 CONFIGURATION......................................................... 167
CHAPTER 11 LCD DISPLA Y CONFIGURATION ................................................................ 174
APPENDIX 1 CDR FORMAT................................................................................................ 177
APPENDIX 2 EXPORTED FILE FORMAT ........................................................................... 179
APPENDIX 3 SIPPBX 6200 STATUS CODE........................................................................ 186
APPENDIX 4 DEBUG LOG TOOL ....................................................................................... 187
APPENDIX 5 BUILD-IN VOICE PROMPT INDEX................................................................ 188
APPENDIX 6 TIME ZONE TO COUNTRY MAPPING LIST................................................. 203
APPENDIX 7 STEP BY STEP SETTING FOR WAKEUP CALL.......................................... 205
APPENDIX 8 STEP BY STEP SETTING FOR CONNECTING TO CAS2000 BILLING
SOFTWARE.......................................................................................................................... 208
APPENDIX 9 SERVICE TYPE LIST......................................................................................211
APPENDIX 10 STEP BY STEP TO CREATE A NEW LANGUAGE OF PROMPT.............. 212
APPENDIX 11 TTS FILE INDEX........................................................................................... 214
APPENDIX 12 STEP BY STEP TO MANAGE LP600N’S FIRMWARE AND SETTINGS
(AUTO CONFIGURA TION)................................................................................................... 220
APPENDIX 13 STEP BY STEP TO MAKE LP600N RUNNING (AUTO CONFIGURATION)
...............................................................................................................................................
APPENDIX 14 LP600N EXAMPLE CONFIGURATION FILE .............................................. 225
222
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Chapter 1 SIPPBX 6200 Series Introduction
The Welltech SIPPBX 6200S/GS/N is an all in one IP-PBX which including
PABX telephony service, Auto Attendant, Voice Mail, Service Setting IVR,
Enterprise Coloring Ring Back Tone, Conference, Announcement, FAX
Service, Toll Restriction and VOIP router features together. It supports up-to
1000 subscribers and is a cost effective solution for small to medium
enterprise. Also the traveler soft-phone, operator console and billing software
provides you a complete transition from traditional PABX to the new generation
IPP-BX.
SIP RFC 3261 Compliance
Support G.711, G.729A, GSM, G.723(optional)
Support SIP/RTP Encryption/Decryption
Support RADIUS Server or Enterprise Billing via TCP
Support High Available Redundant (optional)
Support SIP ENUM
Support VLAN & QOS tagging
Support Setup Wizard
Built-in CDR Report
Auto Attendant
Web-Base Auto Attendant Flow Editor
Scheduled Special Announcement
Holidays Working Time Support
Multiple Language Support
Support Branch Office
Support Transit Call
Voice Mail
Web-Base Voice Mail Flow Editor
Personal Greeting
Multiple Language Support
Native TTS (Chinese & English & Japanese) Support
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Support Additional Customized TTS Language
Message Waiting Indication
Email Notify
Web Retrieve
Phone Retrieve
Service Setting IVR
Web-Base Service Setting Flow Editor
Multiple Language Support
Native TTS (Chinese & English & Japanese) Support
Support Additional Customized TTS Language
Access Key Setting Result Announcement
Conference Bridge
Support RFC 4579 (without XML)
Ad-Hoc Conference
Virtual Conference (Meeting Me)
Virtual Conference (Ad-hoc)
Support Host Creation and Participant Join PIN
Event Tone Notice
Up-to 8 Participants per Room
Quick Conference by Soft-phone (SP362)
Enhanced Service
System Announcement Service
Company-wide Coloring Ring Back Tone Service
Provided Server Hold Tone
Voice Rout er
Public and Private IP Legs
SIP-Aware RTP Routing
Natural VOIP Firewall/NAT
FAX Service
Support T.38/UDPTL/ECM
Support Personal/Common Fax Account
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PDF Format Storage and Retrieve through Web
FAX to Email Notice
Toll Restriction (CDR Mode only)
Provided Office/Extension Level Toll Restriction Service
Support Allow/Disallow/Allow in Duration Call Restriction
Support Restriction Engaged Time (per 30 minutes)
Support Tone Alert before Disconnect (CPE support is required)
Optional Features
Traveler Soft-phone (SP 362)
Operator Console Software
Enterprise Billing Software
Web Caller Module (optional)
Microsoft Live Communicator/Exchange 2007 Module (optional)
Selected Telephony Features
Call Transfer
Call Forward
Call Forwarded Notice
Call Screening
Caller ID Privacy
Call Waiting
Call Hold
Call Pickup (Global, Group)
Specified Call Pickup
Find Me
Short Code
Do Not Disturb
Miss Call Notify by Email
ANI Replacement
Call Return
Hide ANI/Show ANI Selection
Call Park/Retrieve
Call Camp on
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Display Name Replacement
PSTN Number
Ring PSTN & IP Device Simultaneously
Broadcasting Service
Wake-Up Call
Reject Anonymous Call
Support SIP TAPI
Busy Lamp Filed (RFC 4235)
SIPPBX 6200 Appearance Description
In order to satisfy the different customers’ requirement, SIPPBX 6200 has 3
models: SIPPBX 6200N, SIPPBX 6200GS and SIPPBX 6200S.
System Capacity of each model.
Model
Name
Max
Subscriber
Max Current
Call
Max RTP
Resource
DSP channels FAX channels
6200N Up to 1000 Up to 500 Up to 250 Up to 120 Up to 16
6200GS Up to 400 Up to 200 Up to 120 Up to 48 Up to 8
6200S Up to 200 Up to 100 Up to 60 Up to 24 Up to 4
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SIPPBX 6200N Front Panel:
Welltech Computer Co., Ltd.
Functions:
1: Power LED
2: H/D LED
3: System Status LED
4: Network1 Interface LED (not used)
5: Network2 Interface LED (not used)
6: LCD Panel
7: LCD Touch Panel
8: Power Switch
SIPPBX 6200N Rear Panel:
76 8 54 3 2 1
2
3 1
4
5
Functions:
1: Electric Fan
2: AC Power outlet
3: AC Power switch
4: Keyboard/Mouse
5: Console port
6: SIP Service Ethernet port (WAN)
7: Management (Voice Gateway) Ethernet port (LAN)
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6
7
8
9
8: VGA
9: USB (not used)
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SIPPBX 6200GS Front Panel:
2
1
3
4
5
6
Functions:
1: Power LED
2: Network1 Interface LED (not used)
3: Network2 Interface LED (not used)
4: H/D LED
5: Power Switch
6: System Status LED
7: LCD Panel
8: LCD Touch Panel
SIPPBX 6200GS Rear Panel:
1 2 3
7
8
4
5
6
7
8
9
Functions:
1: Electric Fan
2: AC Power outlet
3: AC Power switch
4: Keyboard/Mouse
5: Console port
6: SIP Service Ethernet port (WAN)
7: Management (Voice Gateway) Ethernet port (LAN)
8: VGA
9: USB (not used)
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SIPPBX 6200S Front Panel:
Display the system status of SIPPBX 6200S.
SIPPBX 6200S Rear Panel:
1
2
4
5
Functions:
1: AC Power switch
2: AC Power outlet
3: VGA
4: Console port
5: USB (not used)
6: Keyboard
7: Mouse
8: LAN Management (Voice Gateway) Ethernet port
9: WAN (SIP Service Ethernet port)
8
9
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Chapter 2 Welltech SIP PBX 6200 Quick Start
After connected Ethernet cables into the SIP PBX 6200 Service Interface,
turned on the power. The first step is to logon the system and set up the IP
address.
Before you can use the browser to config SIPPBX 6200, you need to
install Java Plug-in. Please confirm your JRE version is 1.4.2(preferred &
tested), if your PC has already installed Java.
You also need to set newer versions of stored pages in Internet Explorer to
“Every visit to the page.” Click Tool > Internet Option > General > Setting.
After success, restart your browser to take effect.
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Logon SIP PBX 6200
Step1: Start IE6.0 (or later version) to navigate SIPPBX 6200 web
management system by typing the default URL is
http://192.168.111.101:10087. The screen will display User ID and
Password as figure 2.1-1.
Figure 2.1-1
Note: The default network IP address is:
Service Interface: 192.168.111.101/ 255.255.0.0/ 192.168.111.254
Management Interface: 192.168.111.102/ 255.255.0.0
Make sure your PC had same network IP address setting (e.g.
192.168.111.33/255.255.255.0) Otherwise, you will be unable to connect to
SIPPBX 6200.
Step 2: Enter login user name and password (the default user ID is root and
user password is root). You can manage your user account via web
(refer to section Account Manager) later.
Figure 2.1-2
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Step 3: The screen shows the Home Page of SIP PBX 6200 as Figure 2.1-3.
Figure 2.1-3
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Chapter 3 SIPPBX 6200 Setup Wizard
Initial Setup Wizard
This is a wizard to help administrator easily to management their PBX settings.
Start Path: Home > Initial Setup Wizard
This is wizard should be run at first. It will guide the administrator to setup
the basic system parameters step by step.
Step 1: Change the root and admin’s password for security reason and setup
the preferred user interface language.
Figure 3.1-1
Field Description:
• User ID: Login User ID
• Password: Login Password
• Confirm Password: Confirm new password again
• Language: The language of web user interface for the user.
Note: The system provides 2 USER ID by default: root and admin.
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Step 2: Change the system time zone.
Figure 3.1-2
Parameter Description:
• Time Zone:
- Standard: Use a predefined standard time zone(Refer to section
“Timezone to Country Mapping List”)
- Customize: Use a user defined time zone
• Auto Daylight Saving: Auto adjust daylight saving timer or not
Step 3: Change the date and time for the system.
Figure 3.1-3
Parameter Description:
• Date (yyyy/mm/dd): The date for the selected time zone
• Time (hh:mm:ss): The time for the selected time zone
Step 4: Choose to enable SNTP or not. If yes, please setup the SNTP server
and related parameters.
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Figure 3.1-4
Parameter Description:
• SNTP Server: The SNTP server list. You can use; or, to have multiple
SNTP server.
• Polling interval (second): How long the SNTP will synchronize the time from
the SNTP server in seconds.
• Min Retry Interval (second): This is the initial retry interval. Each retry will
multiple 2. For example, the Min Retry Interval is 10 and Max Retry Interval
is 600. The retry sequence will be 10, 20, 40, 80, 160, 320 seconds.
• Max Retry Interval (second): This is the max time to wait before attempting
a retry.
• Max Adjust Time (second): If this is zero, the SNTP client will be willing to
apply any size time adjustment. If this is non-zero, then any time
adjustment greater than this will be considered an error.
• NTP Server Port: This is the port number the SNTP client will attempt to
connect to on the time server. The default value is 123.
• Local NTP Port: This is the port number that will use to listen for replies
from the server. The default value of 0 means use a random port number.
• Packet Error Check: If this is disabled, then the validity checks are
suppressed. The default value is enabled.
Step 5: Choose to enable the voice gateway mode or not.
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Figure 3.1-5
Parameter Description:
• Voice Gateway: Enabling voice gateway feature, SIPPBX 6200 will be able
to play the role as a NAT server to pass through SIP and voice call. Please
refer to Voice Gateway Example
for a configuration example.
Note: Enable Voice Gateway will require 2 Ethernet ports connected, one is for
WAN and another is for LAN (private network). It can play as a SIP-aware
voice NAT router. User can either register to LAN interface or WAN
interface. NAT transversal will be done automatically in between.
Step 6: Setup the corresponding network parameters.
Figure 3.1-6
Parameter Description:
WAN Interface
• IP Address: The server IP address (WAN)
• IP Netmask: The server IP netmask (WAN)
• IP Gateway: The server default gateway
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• Primary DNS Server: Primary DNS Server IP network
• Secondary DNS Server: Secondary DNS Server IP network
• Host Name: Host name used to register to DNS Server
• Domain Name: Domain name used to
• Dynamic DNS Registration: Enable Dynamic DNS registration or not
LAN Interface
• IP Address: The server IP address (LAN)
• IP Netmask: The server IP netmask (LAN)
• Enable VLAN: Enable the virtual LAN or not
• VLAN ID: The VLAN ID
Note: If you are using voice gateway mode, please make sure you have
connected both Ethernet ports.
System Setup Wizard
After completing the initial setup, you need set the basic parameters for
the server system. For example, UDP port, Enable the Encryption or not, and
so on. Here is the second step required after initial setup wizard.
Start Path: Home >System Setup Wizard
Step 1: System Setup Basic Parameters. SIPPBX 6200 can support up-to 3
UDP/TCP ports. If you are using voice gateway mode, please make
sure you have connected both Ethernet ports.
Figure 3.2-1
Parameter Description:
• SIP Domain: WellSIP Telephony SIP Proxy domain name. It’s normally
used when you have a DNS record setup for WellSIP 6500.
• Listen UDP Port , 2 and 3: The local UDP port on which the SIP service
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listened
- Encrypt: The SIP signal and voice RTP will be encrypted while passing
through the UDP port.
• Enhance Service Port : The listen UDP port for enhance service
• MWI Service Port: The listen UDP Port for MWI Service (8080-8085)
• Subscriber Login: Enable Subscriber login to WellSIP 6500 or not
• Support Video: Support video RTP proxying or not. Enable video will great
reduce the number of concurrent RTP channel and bandwidth.
• Forward Caller ID:
- Caller: use original caller ID when call is forwarded
- Forwarder: use forwarder caller ID when call is forwarded to another
user
• Voice Gateway: Enabling voice gateway feature, WellSIP 6500 will be able
to play the role as a NAT server to pass through SIP and voice call. Please
refer to Voice Gateway Example for a configuration example.
Step 2: Setup system event email notice and voice mail to email parameters.
Figure 3.2-2
Parameter Description:
System Event Notice
• Enable System Log: Enable to send system information to syslogd
Server or not
• SyslogD Server IP 1, 2: syslogd server IP address
• SNMP Sending Interface: The SNMP sending interface
• Event Email Notice: Enable the event email notice or not
• SMTP Event Filter Level: The level of filter
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• SMTP Mail Server: SMTP server host for email notice
• Auth Mode: The authentication type for the SMTP server
• Email From: Email sender account
• Email To: Email receiver (semicolon is used for multiple receiver)
• Subject: Email subject to be send to receiver. The following variable
parameters can be used to create dynamic subject for system notice:
- $LOGLEVEL$: Information Level
- $HOSTNAME$: Host name
- $HOSTIP$: Host IP address
VMS Notice
• Enable Notify: Enable email notice or not
• SMTP Mail Server: SMTP Mail Server
• Auth Mode: Authentication mode, none, cram MD5,login,plain or NTLM
• From Address Automatic Assign: Enable from address automatic assign or
not. If it is set to automatic, the email From address will be
Telno@SIPPBX_IP.
• Manual Assign Address: Manual assign the FROM mail address
• Email Subject: Email subject. The following variable can be used for VMS
Email notice:
$ANI$: Calling Number
$MSG_TIME$: The voice mail message date and time
$MSG_FILE$: The voice mail message file name
• MP3 Attach: attach MP3 file or not
Step 3: Setup the corresponding service prefix and service account
parameters according your dialing/number plan.
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Figure 3.2-3
Parameter Description:
• Enhance Service Account: Service account of enhance service. The
system will automatically add the account you inputted here.
• MWI Service Account: Service account of MWI service. The system will
automatically add the account you inputted here.
• Server Broadcast Prefix: The prefix of the broadcast. The broadcaster uses
this prefix + broadcast-group-id to select which broadcasting group will be
used. 6200 will initiate all calls out-to broadcasted number. It is required to
have the specified CPE device to understand the broadcasting server
request. SIPPBX 6200 can allow having up-to 16 parties of current
broadcasting target for 6200s and 30 parties for 6200. The system wide
total broadcasting target is same as DSP channel license.
• Service Setting IVR Prefix: This is the prefix to be used for Service setting
IVR. The call flow of Service Setting IVR can be customize by drag and
drop with call flow editor. User can dialing in this prefix and IVR will guide
for setting the service.
• VMS Access Code 1: Use extension number as the VMS user to access
voice mail. No extension input is required.
• VMS Access Code 2: Will ask for extension number and password
authentication to access VMS.
• Leave VMS Directly Prefix: This is the prefix to be used for entering the
leaving message directly without prompt. The system will automatically
create the required prefix. The users only need to dial this prefix and
extension number for leaving the message. It is useful for operation
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console to transfer the customer into a voice mail.
• Conference Prefix: Conference Service URI
• Virtual Conference Prefix: Virtual conference prefix code
• Wakeup Call Prefix: Wakeup call service prefix code
• FAX Prefix: The prefix of the FAX Service. The Extension can receive the
FAX service by following ways.
1. FAX in through Auto Attendant: The Caller dials into Auto Attend first
then transfer to “FAX Prefix+Extension” and press start fax of the fax
machine.
2. Direct Fax in number: Create a new extension as the fax DID number
and enable unconditional forward to “FAX Prefix+Extension”.
The Caller can send fax by dialing this fax number directly.
The received FAX messages will be stored as PDF format.
The fax message can be retrieved through web interface or send a email
to the user with the fax PDF attachment.
Step 4: Choose the minimum reserved resource according to your license.
Figure 3.2-4
Parameter Description:
• Reserved Auto Attendant & VMS Channels: The min channels reserved for
the auto attendant and VMS
• Reserved conference Channels: Min conference channels to be used
• Reserved CRBT/Announcement Channels: Min announcement/CRBT
channels to be used
• Min Wakeup Call Channels: Minimum wakeup call channel will be used
(reserved) during the wake up call service.
• Max Wakeup Call Channels: Maximum wakeup call channel will be used
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for wakeup call service. Over this limit, the rest of request will be queued.
• Reserved Service Setting Channels: Minimum Service Setting channel will
be used for Subscriber service setting IVR or announcement
Headquarter/Branch Office Setup Wizard
In this wizard, you can define each profile for offices.
Start Path: Home > Headquarter/Branch Office Setup Wizard
Step 1: Add or modify headquarter or branch office profile here. The figure is
as figure 3.3-1.
Figure 3.3-1
Parameter Description:
• New or Modify: New a office or modify the existing office
• Office ID: The branch office ID
• Name: The branch office name
• Office Phone Number: The incoming telephone number of the branch office
• AA Primary Language : The auto attendant preferred language
• AA Secondary language: The auto attendant second language
• DM Group ID: Group-wide digit manipulation applied
Step 2: Choose the office's time zone.
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Figure 3.3-2
Parameter Description:
• Office ID: The branch office ID
• Time Zone: The branch office time zone
- Standard: Use a predefined standard time zone (Refer to section
“Timezone to Country Mapping List” )
- Customize: Use a user defined time zone
• Auto Daylight Saving: Auto adjust daylight saving time or not
Step 3: Setup the service access code according to your number plan. Please
don't conflict with your extension number.
Figure 3.3-3
Parameter Description:
• Service Type: Applied service type
• Service Code: Telephony Keypad used for the service code
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Step 4: Setup the office's holiday, work time and lunch time parameters.
Figure 3.3-4
Parameter Description:
• Holiday: Non-working weekday
• Office Hour: The working time setting for a week
• Lunch Break: The lunch time setting for a week
• SUN ~ SAT: The week day
•
Step 5: Update your holiday information for the selected office.
Figure 3.3-5
Parameter Description:
• Date: Holiday date, the format is MM/DD.
• Description: The description
Note: In this version, all subscribers either belong to headquarter or belong to
the branch office. There are not the subscribers who don’t belong to the
office.
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Extension Setup Wizard
After you define the proper office profile and in here you can setup the required
extension.
Start Path: Home > Extension Setup Wizard
Step 1: Add or modify the extension number for the office.
Figure 3.4-1
Parameter Description:
• New or Modify: You want to new or modify an extension
• Office ID: Belonged office ID
• Department: Belonged department ID
• Device Type: Subscriber device type
- Subscriber: Subscriber user
- Softphone: It is Welltech soft-phone device type. Welltech SP-365 is
required as the soft-phone. When you define an extension to a
softphone, it means that you can use this account for SP365 or regular
SIP devices.
• Subscriber ID: The subscriber ID
Step 2: Provides some basic parameters such as pickup group assignment,
password etc.
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Figure 3.4-2
Parameter Description:
• Pickup Group ID: Pickup group ID for per subscriber
• User Password: Register user password (device password only)
• Web Password: Password used for web access
• No Answer Timer: The maximum time (in second) to wait the remote party
answer (pick up phone).
• Description: The description
Step 3: Setup those call related service parameters such voice mail, forward
etc.
Figure 3.4-3
Parameter Description:
• Unconditional Forward: When enabled, any calls to this subscriber will be
forward to this URI unconditionally. You can use SIP URI or subscriber ID
here.
• Display Name: Change SIP display name to the set name.
• No Answer Forward: Forward to the URI when the subscriber has no
answer.
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• Busy Forward: Forward to the URI when the subscriber is busy. You can
use SIP URI or subscriber ID here. You can use SIP URI or subscriber ID
here
• Unavailable Forward: Forward to the URI when the subscriber is
unavailable (not registered). You can use SIP URI or subscriber ID here.
You can use SIP URI or subscriber ID here
• Group Pickup (Picked): Allowed to be picked-up within group or not
• Global Pickup (Picked): Allowed to be picked-up globally or not
• VMS: Voice Mail Service
Step 4: Here you can setup the conference, call transit etc parameters.
Figure 3.4-4
Parameter Description:
• Virtual Conference Active Mode: To enable the virtual conference service or
not.
• Virtual Conference Room: Virtual conference room number. The final virtual
conference room number is prefix + inputted number.
• Default VMS Language: The default VMS language
• VMS Personal Greeting: Enable personal greeting or not
• Notice Email Address: Email address for the VMS service
• Transit Call: Allow to call to PSTN number inside the Auto Attendant or not.
The administrator needs to add transit call component in auto attendant call
flow editor. The sample call flow can be downloaded from Welltech support
web site.
PSTN Gateway Setup Wizard
The PSTN gateway is used to setup the FXO, E1/T1 or SIP Gateway in order
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to call out to PSTN world.
Start Path: Home > PSTN Gateway Setup Wizard
Step 1: Here you can setup the basic parameters such as register number,
telephone etc for the selected PSTN gateway.
Figure 3.5-1
Parameter Description:
• New or Modify: New or modify a gateway
• Belonged Office: Belonged office ID
• Register Number: Register TEL no or user account
• Register User Password: Register user password (device password only)
• Register Type: Subscriber register type
- Dynamic: Subscriber need send register message for availability
- Predefine: Subscriber will be handle as a permanent user
Predefine URI1: Predefine subscriber URI1 (i.e.
sip:9001@203.69.216.53)
Predefine URI2: Predefine subscriber URI2 (i.e.
sip:9001@203.69.216.54)
- Predefine/NAT: Subscriber will be handled as a permanent NAT user
(manual IP/Port mapping is required)
Predefine URI 1: Predefine NAT subscriber URI (i.e.
sip:8001@192.168.19.61: 7777)
Public TA: mapped NAT Server IP address and port (i.e.
61.218.42.217:5060)
- LCS Dynamic: This is used only when the subscriber type is a regular
subscriber and also a LCS user. You can use IP phone or CPE to
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register this account and also it will ring the LCS communicator client
software simultaneously.
• Device Type: Gateway device type
- Gateway: Gateway (e.g. trunk gateway or FXO gateway)
- Gateway/RTP: Welltech WG5250/WG5200
- Proxy/RTP: Welltech SIPPBX 6200
- SIP Proxy: SIP proxy server
- IVR/VMS: IP IVR or VMS server
- IVR/VMS/RTP: Welltech IVR or VMS server
- Recorder: Welltech Recorder
- Outbound Caller: Outbound Caller
- Register UAC: Register user agent client
User Agent ID: User agnet ID in UAC
Caller Info: Display calling parting information
▪ Caller TEL No: Display original caller ID
▪ Registered TEL No: Registered UAC user ID
▪ Caller Display Name: SIP display name for original caller
- LCS Server: Microsoft LCS Server
- MWI Server: SIP Message Waiting Indicator Server
- Presence Server: Presence Server
- RTP Server: RTP Server
- VMS (Diversion): The voice mail server which support diversion header
as draft-levy-sip-diversion-08.txt.
- Web Caller: It is used for Welltech Web call service. Welltech will
provide OCX and sample code for integrating into the customer’s Web
server (Web call license is required). It can be used for click to call or
web 800 calls.
- Exchange Fax: Allow to provide the fax feature for Exchange 2007.
Regular T.38 device can work with Exchange 2007 UMS when using
this exchange fax user (Additional license is required).
- Diversion/VMS/RTP: This device type is used only for Welltech IPcentrx
6850. It will be able to greatly reduce the usage of RTP resource.
- Gateway/RTP/Fax: This device type is dedicated for Wellgate 5260.
• Predefine IP: The predefine server IP address
• Predefine Port: The predefine server UDP port
• Description: The description
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Step 2: Choose the prefix you are going to add or modify for adding or
removing the selected gateway.
Figure 3.5-2
Parameter Description:
• Add into exist call routing: Add the following prefix into the call routing
existed
• Create a new call routing: Create a new call routing for the following prefix
• Prefix Matched: Called number prefix to be matched
• TEL NO: Subscriber TEL no for route
• Priority: Used only for priority hunting.
Note: After you create the PSTN Gateway information successfully, you will
see the corresponding record in the Gateway part and Call Routing part.
SIP Trunk Setup Wizard
Instead of you can peer to peer to connect to a SIP service provider. The
SIP service provider might give you a SIP account for register and making call.
For this kind of configuration, you need to use the SIP trunk setup here.
Start Path: Home > SIP Trunk Setup Wizard
Step 1: Set the SIP trunk parameters which provided by your SIP service
provider.
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Figure 3.6-1
Parameter Description:
• Office: The office ID belonged
• TEL NO: The office TEL NO
• Description: The description
• User Agent ID: Identifier used for subscriber setting (type UAC)
• Register TEL No: SIP registrar telephone number
• Register ID: SIP registrar user ID
• Register Password: SIP registrar user password
• SIP Realm: SIP registrar domain
• Register IP: SIP registrar IP address
• Register Port: SIP registrar UDP port number
• Register TTL: The registration maximum time to live setting when
registered to the SIP registrar
• Use Outbound Proxy: Use the SIP outbound proxy server or not
Step 2: Choose the prefix you are going to add or modify for adding or
removing the selected SIP trunk.
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Figure 3.6-2
Parameter Description:
• Add into exist prefix: Add the following prefix into the call routing existed
• Create a Prefix Route: Create a new call routing for the following prefix
• Prefix Matched: Called number prefix to be matched
• TEL NO: Subscriber TEL no for route
• Priority: Used only for priority hunting.
Note: After you create the SIP trunk successfully, you will see the
corresponding records in the UAC part and the Call Routing part.
Call Pickup Group Wizard
Within the same pickup group, the users are allowed dialing the group pickup
access code to pickup a call within the same group.
Start Path: Home > Call Pickup Group Wizard
Step 1: Choose or add a pickup group for management.
Parameter Description:
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• Office: The office ID
• Pickup Group ID: The pickup group belonged
• Description: The description
Step 2: Manage your group member for the selected pickup group.
Figure 3.7-2
Parameter Description:
• Extension: The extension number belonged to this pickup group ID
Outgoing Called Number Screening Wizard
You might need to limit the users’ outgoing call, such as long distance,
international calls etc.
Start Path: Home > Outgoing Called Number Screening Wizard
Step 1: Add/modify a called number screening group.
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Figure 3.8-1
Parameter Description:
• Group ID: The screening group ID
• Description: The description
Step 2: Manage your outgoing called screening number for the selected
group.
Figure 3.8-2
Parameter Description:
• Screening Prefix: The screening prefix for the outgoing call
• Screening Type: The screening type, allow or disallow.
Step 3: Select office for group members which will applied to the group.
Figure 3.8-3
Parameter Description:
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• Select Office: Select the office for group members which apply the
screening group
Step 4: Add or remove group members which will applied to the selected
group.
Figure 3.8-4
Parameter Description:
• Extension & Gateway: Display all the extension and gateway number
• Applied Extension & Gateway: the extension and gateway number that will
be applied to the screening group
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Chapter 4 System Core Reference
All system related parameters can be set here.
System
Start Path: System Core > System
Figure 4.1-1
Parameter Description:
• SIP Domain: SIPPBX SIP domain name. It’s normally used when you have
a DNS record setup for SIPPBX 6200.
• Listen UDP Port, 2 and 3: The local UDP port on which the SIP service
listened. When you check the encrypt option, this SIP license port will use
Welltech proprietary encryption SIP/RTP for ISR blocking or security. When
an encrypted device talk to un-encrypted device, 6200 will reroute the RTP
back to server to make them talking.
• TCP Enabled: Enable the local TCP port or not
• Local TCP Port, 2 and 3: The local TCP port on which the SIP service
listened
• No Answer Timeout: The default maximum time (in second) to wait the
remote party Answer (pick up phone).
• Max Forward Times: The maximum times to forward the calls
• Default Max Register Time: The default maximum register for public
network user when a subscriber user is crated
• Default NAT Max Register Time: The default maximum register time for a
inside user when a default subscriber user is crated
• Enable Device ACL: Authenticate specified device type or not.
• First Response Timeout: The default maximum time to wait for response.
It’s depended on the network speed.
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• Subscriber Login: Enable Subscriber login to SIPPBX 6200 or not
• Over Max Contact Rule: Over Max Contact Rule, reject or update.
- Reject: The system will reject the new contact REGISTER request
when the subscriber’s used contacts reached the max contact
- Update: The system will replace the oldest contact by new received
contact.
• Support Video: Support video RTP proxy or not. Enable video will greatly
reduce the number of concurrent RTP channel and bandwidth.
• Voice Gateway: Enabling voice gateway feature, SIPPBX 6200 will be able
to play the role as a NAT server to pass through SIP and voice call. Please
refer to Voice Gateway Example for a configuration example.
• CPE Billing Enquiry: Enable the CPE billing enquiry or not. It requires
WellBilling 6600 and Welltech CPE device for this service.
• Forward Caller ID:
- Caller: use original caller ID when call is forwarded
- Forwarder: use forwarder caller ID when call is forwarded to another
user
• Accept Anonymous Call to Subscriber: Whether to allow the anonymous
caller to dial to the subscriber. It is mainly used when ENUM support is
enabled.
• Enable ENUM: Enable the ENUM or not. It is required to setup the DNS
server in SIP service interface and input the correct ENUM Domain Suffix.
• QOS Type: Quality of Service Type
- None: Not using QOS Tag
- DiffServ: Differentiated Services Value
- TOS: Type of Service.
Advance System Configuration
Start Path: System Core > System > Advance
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Figure 4.1-2
Advance Parameter Description:
• Call Validation Time: The timer to check periodically call is valid or not.
SIPPBX 6200 will send re-INVITE periodically for call validation. This
parameter also refers to the session timer RFC 4028 Min-se header. 422
will be sent when Session-Expires is smaller call validation time.
• NAT Compare Method: How to detect a NAT user
- IP Only: Compare IP only
- IP / Port: Compare IP and UDP port
• RetransmissionT1 (msec.): T1 determines several timers as defined in
RFC3261. For example, when an unreliable transport protocol is used, a
Client Invite transaction retransmits requests at an interval that start at T1
seconds and doubles after every retransmission. A Client General
transaction retransmits requests at an interval that starts at T1 and doubles
until it reaches T2. (Default Value: 500ms) **
• RetransmissionT2 (msec.): Determines the maximum retransmission
interval as defined in RFC3261. For example, when an unreliable transport
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protocol is used, general requests are retransmitted at an interval which
starts at T1 and doubles until reaches T2. If a provisional response is
received, retransmission continue but at an interval of T2. (Default Value:
4000ms) **
• RetransmissionT4 (msec.): T4 represents the amount of time the network
takes to clear message between client and server transactions as defined
in RFC3261. For example, when working with an unreliable transport
protocol, T4 determines the time that UAS waits after receiving an ACK
message and before terminating the transaction. (Default Value: 5000) **
• Cancel General No Response Timer (msec.): When sending a CANCEL
request on a General transaction, the User Agent waits cancel
GeneralNoResponseTimer milliseconds before timeout termination if there
is no response for the cancelled transaction(Default Value: 10000ms).**
• General Request Timeout Timer (msec.): After sending a General request,
the User Agent waits for a final response generalRequestTimeoutTimer
milliseconds before timeout termination (in this time the User Agent
retransmits the request every T1, 2*T1,…T2,…milliseconds)**
• Proxy 2xx Rcvd Timer (msec.): A successful client INVITE transaction of a
Proxy server includes only the INVITE request and the 2xx response. (The
ACK is not part of the transaction.)After receiving the 2xx response, the
Proxy will wait proxy2xxRcvdTimer before the transaction terminates.
(default: 10000 ms)**
• Proxy 2xx Sent Timer (msec.): A successful server INVITE transaction of a
Proxy server includes only the INVITE request and the 2xx response. (The
ACK is not part of the transaction).After sending the 2xx response the
Proxy will wait proxy2xxSentTimer before the transaction will terminate.
(default: 8000 ms)**
• Use Domain for Auth: Send “Domain” in 401 or 407 for authentication or not
• General Guard Time: The general guard time for internal purpose only
• Nonce Valid Period: The max valid time for a nonce. Once time out,
SIPPBX 6200 will issue a new nonce for authentication. Set it to 0 will
cause SIPPBX 6200 to generate new nonce for each call or register
• Valid Period Auth Mode: During the nonce valid period, does a subscriber
need send response on the “register”/“invite” message or not
- None: User agent doesn’t need send MD response in register invite
message
- MD: User agent should send MD response over the current nonce. Or a
new nonce will be send by SIPPBX 6200
• Message Pool Page Size: Used to hold and process all incoming and
outgoing in the form of encoded message or message objects. It is
recommended that you configuration the page size to the average
message size your system is expected to manage.
• General Pool Page Size: Used by SIP Stack objects, such as call-legs and
transactions, to store the internal fields. For example, the call-leg object will
store the To, Form and Call-ID headers and the local and remote contact
addresses on the general pool pages. The general pool is also used for
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other activates that demand memory allocation.
• Send Receive Buffer Size: The buffer size used by the SIP Stack for
receiving and sending SIP messages.
• Memory Pages: Number of memory page allocated.
• Support Video: Support Video or not. Enable video will greatly decrease the
number of concurrent channel and bandwidth.
• RTP Resource Timeout: The maximum time to wait for RTP server
response. It’s depended on the network speed. (only available when
working with Welltech external RTP resource server)
• System Announcement: Used when personal announcement cannot be
located (e.g. user not found).
• Invalid TTL Process: Response policy when register expired is too small.
- Use Proxy TTL: Response proxy expires time to UAC and expect it will
use it as default TTL.
- Reject: Send 423 Interval Too Brief to UAC
• AAA Sending Stage: Send AAA message before or after DM
• Global Call Validation: Call Validation through both parties, none, caller or
called
• Use Local Time: Enable use local time or not
• Missed Call From Domain: Domain of missed call, use none, IP address or
domain name
• Missed Call Tel No: Telephone number of missed call, use TEL No or
replace_ANI
• Camp on Timeout: When user camp on a call, the maximum time to wait
before called extension to become free.
• Web Call: Enable to use demo web call for subscriber or not. It requires the
web call license from Welltech.
• Web Call ID: The demo web caller ID, you can get a number by click .
• Camp On Codec: The codec will be used for a camp on calls. It is required
to use a most of common codec within your PABX.
• Change Call-ID: Whether change the call-ID in the SIP message when
forking a call out. Please contact Welltech Engineer when you want to
change it.
• Dedicate UAC Failure: It is used when you are using dedicated UAC
feature. If a subscriber is using the dedicate UAC ID and it is not registered,
this option will decide which actions will be taken. Select “Local Prefix
Routing” when you would like to use backup route. Select “Disconnect”
when you are not allowed using backup route.
• BLF (RFC 4235) State Code: Busy Lamp Field State Code when report the
subscriber status to Subscriber.
- Unavailable: The default is “void”.
- Idle: The default is “terminated”.
- Ringing: The default is “early”.
• Subscriber server-based 302: Enable Server based 302 Move or not.
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When it is enabling, the 302 will be processed by SIPPBX 6200 instead of
sending it to caller.
** SIP and network knowledge is required to change these parameters.
Debug
Debug can be turn on or off based on each system module and level to
minimum the debug information. Please only turn on the debug information
for debug purpose under Welltech FAE's instruction and turn off when
complete. Or the system performance will be greatly hit.
Start Path: System Core > Debug
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Figure 4.2-1
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License
Start Path: System Core> License
Figure 4.3-1
License Parameter Description:
• Feature: System parameter
• Serial No: System parameter
• License Key: System parameter
Note: Please don’t change it unless under Welltech’s instruction.
CDR
The system can support CDR more for connecting to an enterprise billing
server such as CAS 2000. The Appendix 8 descripts the step by step setting
for connecting to CAS 2000 billing software. The below is the CDR
parameters.
Start Path: Configuration > CDR
Figure 4.4-1
Parameter Description:
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• TCP Port: The local TCP port
• Trust Provisional IP: Trust provisional IP or not
• Auth Mode: Enable log in request or not
• Login Prompt: Login prompt
• Password Prompt: Password prompt
• CDR Format: CDR format. Please refer to Appendix 1 CDR Format
• Max CDR File Keep Days: The maximum CDR system keeping days
• CDR Send Delay (ms): The delay time of send CDR
• Output Mode: The output mode for the CDR. If you need to connect to
External Enterprise Billing system, it is required to have TCP enabled.
• Send Zero Session Time: Enable send zero duration CDR or not. If it is
enabled, the unconnected call will be write to CDR.
• Send Billing Start: Whether to send a start CDR to external CDR server or
not when the call is connected.
• CDR Report: Enable or disable CDR Report. When enable the CDR
reporting, each call will be recorded for reporting purpose. If you are using
the external billing software, it is recommend to not turn it on.
• CDR Report Time Format: The date time format of CDR report. Supported
format show as below,
"mm.dd.yy hh:mm:ss"
"mm/dd/yy hh:mm:ss"
"mm-dd-yy hh:mm:ss"
"mm.dd.yyyy hh:mm:ss"
"mm/dd/yyyy hh:mm:ss"
"mm-dd-yyyy hh:mm:ss"
"yyyy.mm.dd hh:mm:ss"
"yyyy/mm/dd hh:mm:ss"
"yyyy-mm-dd hh:mm:ss"
• Max CDR Report Raw Data Keep Days: The maximum keep days for the
CDR report raw data
• Toll Restriction: Enable or disable the Toll Restriction Service or not. After
enable toll restriction, the administrator will able to restrict each extension’s
call by called prefix or calling time.
• Inter-Subscriber Billing: Enable Billing Report to include Extension to
Extension calls or not.
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AAA
When the subscriber users do the AAA (Authorization, Authentication
and Accounting), enter the correct parameter the Radius setting.
Start Path: System Core > CDR > AAA
Figure 4.4-2
Parameter Description:
• Authorization IP: Radius Authentication/Authorization Server IP address
• Authorization Port: Radius Authentication/Authorization Server Port
• Accounting IP: Radius Account Server IP address
• Accounting Port: Radius Account Server Port
• Backup Authorization IP: Backup Radius Authentication/Authorization
Server IP address
• Backup Authorization Port: Back Radius Authentication/Authorization
Server Port
• Backup Accounting IP: Back Radius Account Server IP address
• Backup Accounting Port: Back Radius Account Server Port
• Max Retry: The maximum retry times
• Response Timeout (msec): The maximum wait for response time from
RADIUS Server
• Switch Threshold: Switch to alternate RADIUS Server when failures are
occurred more than switch threshold.
• Local Port: The local port used for RADIUS Client
• Secret Key: The shared secret key with RADIUS Server
• CISCO Mode:
- Yes: Use Cisco RADIUS mode (have redundant string in vender
attribute)
- No: no
• CDR Mode:
- Enable: Log CDR into the file
- Disable: no
• CDR Keeper Days: CDR system keeping days
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• Vendor ID: RADIUS vender attribute’s vender ID.(Default is 9)
• Send Zero Session Time:
- Yes: Send 0-balance session time for RADIUS when the call failed
- No: no
• Welltech extend RADIUS attrib: Enable the Welltech extended RADIUS
attributes or not. Please only enable this extended attributes when your
WellBilling 6600 is Release 2.0 and you need use Least Cost Routing or
RADIUS routing features.
• Billing Message: Send RADIUS billing message out
• Inter-Subscriber RADIUS Authentication:
- Yes: When a subscriber is calling another subscriber, SIPPBX 6200 will
send RADIUS for call permission
- No: When a subscriber calling another subscriber, SIPPBX 6200 will
not send RADIUS for call permission
• CDR Report: Enable or disable CDR Report. When enable the CDR
reporting, each call will be recorded for reporting purpose. If you are using
the external billing software, it is recommend to not turn it on.
• CDR Report Time Format: The date time format of CDR report. Supported
format show as below,
"mm.dd.yy hh:mm:ss"
"mm/dd/yy hh:mm:ss"
"mm-dd-yy hh:mm:ss"
"mm.dd.yyyy hh:mm:ss"
"mm/dd/yyyy hh:mm:ss"
"mm-dd-yyyy hh:mm:ss"
"yyyy.mm.dd hh:mm:ss"
"yyyy/mm/dd hh:mm:ss"
"yyyy-mm-dd hh:mm:ss"
• Max CDR Report Raw Data Keep Days: The maximum keep days for the
CDR report raw data
HTTPS Certificate
Secure Hypertext Transfer Protocol Certificate, it can enhance the web
page security. The user can put their owned HTTPS SSL certificate here.
Start Path: System Core > HTTPS Certificate
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Parameter Description:
Welltech Computer Co., Ltd.
Figure 4.5-1
• Key Store File Name: The encrypted
certificate file name in SSL layer
• Key Store Password: The password for the SSL certificate. It required t
gotten from the SSL certificate company.
Event Notice
Server event notic
Start Path: System Core > Event
e method setup
Notice
o be
Figure 4.6-1
Parameter Description:
• Enable System Log: En
able to send system information to syslogd
Server or not
• 1, 2: syslogd server IP address
SyslogD Server IP
• SNMP Sending Interface: The SNMP sending interf
ace
• Event Email Notice: Enable the event email notice or no
• SMTP Event Filter Level: The level of filter
• SMTP Server: SMTP server host for email n
otice
• Auth Mode: The authentication type for the SMTP server
• Email From: Email sender account
• Email To: Email receiver (semicolon
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• Subject: Email subject to be send to receiver. The following variable
parameters can be used to create dynamic subject for system notice:
- $LOGLEVEL$: Information Level
- $HOSTNAME$: Host name
- $HOSTIP$: Host IP address
Backup/ Restore
Backup/ Restore pro
configuration here.
Backup Configur
This will backup the configuration including t
Step 1: To backup the running configuration including th
Backup Configuration (With License) to backup to local hard disk a
figure 4.7-1.
vides a way to backup and restore the working
ation (With License)
he license.
e license, click the
s
Figure 4.7-1
Step 2: The whole running configuration will be compress into a zip file (file
name: PBXBak.zip) and transfer back to local as figure 4.7-2.
Figure 4.7-2
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Backup Configuration (Witho
ut License)
Please follow the step 1and 2 to do the backup c
onfiguration (Without License).
Only this backup file will not include the license information.
Restore Configuration
Step 1: Click Restore Con
click Apply button to restore the configuration to the working
configuration as figure 4.7-3.
figuration Select backup file (i.e. c:\export.zip) and
Figure 4.7-3
Compact Database
To make the database
Step 2: Click Compact to compact the
space smaller.
configuration database and Apply
button to compact the database as figure 4.7-8.
Figure 4.7-8
Note: It is need to restart the system to take effect of the Compact
action.
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Chapter 5 Service Reference
Call Routing
SIPPBX 6200 Call Routing can provide prefix hunting base on priority, max
idle time or round robin method. SIPPBX 6200 will use prefix routing plan to do
the corresponding routing. The routing target can be a UAC (register client),
another proxy, gateway or subscribers....etc. Routing policy is defined here.
Start Path: Service > Call Routing
Figure 5.1-1
Click the Modify button:
Figure 5.1-2
Parameter Description:
• Active Mode: The prefix group is active or inactive
• Prefix Matched: Called number prefix to be matched
• Description: Description
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• Matched Length: Applied only when specified length of DINS is matched.
Zero (0) indicate ignore length option.
• Matched User Group: Applied only for specified user group. Others group
will not be applied.
• Hunting Method: Hunting method used for this group
- Round Robin: Call is hunting rotationally until user answer
- Priority: Call is hunting base on priority set until user answer
- Max Idle Time: Max idle one will be hunt first until user answer
- Ring All (First Answer): Send request to all members. When a user
pickup the phone, cancel the others request.
- Round Robin (Ring Only): Send request based on round robin member
selection. Stop hunting when a user response ringing.
- Priority (Ring Only): Send request based on member's priority. Stop
hunting when a user response ringing.
- Max Idle Time (Ring Only): Send request base most idle policy. Stop
hunting when a user response ringing.
- Round Robin (Load Balance): Send request based on round robin
member selection. Stop hunting when return code is defined in the Load
Balance Reason Code.
- Priority (Load Balance): Send request based on member's priority. Stop
hunting when return code is defined in the Load Balance Reason Code.
- Max Idle Time (Load Balance): Send request base most idle policy. Stop
hunting when return code is defined in the Load Balance Reason Code.
- RADIUS Route: Let the RADIUS to decide the routing. It is mainly used
for least cost routing or QOS routing. It requires using WellBilling 6600
Release 2.0.
- ENUM: This routing is use ENUM DNS server to decide the routing. You
have to set the corresponding ENUM DNS in the Networking settings. If
you are trying to accept ENUM incoming call, you need also enable the
Accept Anonymous Call to Subscriber in system configuration.
Otherwise, the call from ENUM (unknown host) will be rejected.
- No Answer Timeout: The maximum time (in second) to wait the remote
party answer (pick up phone)
• First Response Timeout: The maximum time to wait for device response.
It’s depended on the network speed.
• Remove Prefix: Remove prefix matched or not
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• RADIUS Authorization Resend: Whether to send RADIUS authorization
when doing next hunting or not
Click Detail button to define member of call routing group:
Figure 5.1-3
Parameter Description:
• TEL NO: Subscriber TEL No for route
• Priority: Used only for priority hunting
Click the Reason button:
Figure 5.1-4
Parameter Description:
• State Code: State code for Load Balance reason. When the SIP return
coded is defined here, the load balance hunting will be stopped.
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Digit Manipulation
SIPPBX 6200 Digit Manipulation can provide operator target called
number and calling number to “insert, replace or drop”.
Start Path: Service > Digit Manipulation
Figure 5.2-1
Click the Detail button:
Figure 5.2-2
Modify Digit Manipulation:
System will only execute 1 ANI DM and 1 DNIS DM for a call. Matched
ANI DM will be executed first and use the result for DNIS DM. Click the Modify
button:
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Figure 5.2-3
Parameter Description:
• Group ID: Group ID number
• Matched Prefix: Calling/Called number party matched
• Matched Target: Matched target is ANI(calling number) or DNIS(called
number)
• OP Target: Operator target is ANI(calling number) or DNIS(called
number)
• Matched Length: The matched target length
• Apply Target: The target to be applied
• Active Mode: The DM group is active or inactive
• Start Position: Start position to be replaced
• Stop Position: Stop position to be replaced
• Replace Value: Replaced value
Example 1:
Matched
Prefix
02 DNIS ANI
* DNIS ANI
Matched
Target
OP
Target
Before DM
DNIS: 0282265699
ANI: 070700001
DNIS: *282265699
ANI: 070700001
Start
Position
1 1 6666
0 15 12345
Stop
Position
Replace
Value
DNIS: 0282265699
ANI:06666
DNIS: *282265699
After DM
70700001
ANI:12345
0 DNIS DNIS
0707 ANI ANI
07070 ANI DNIS
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DNIS: 0282265699
ANI: 070700001
DNIS:0282265699
ANI: 070700001
DNIS: 0282265699
ANI: 070700001
0 0 9
15 15 3333
3 5 7777
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DNIS:90282265699
ANI: 070700001
DNIS: 0282265699
ANI: 0707000013333
DNIS: 028777765699
ANI: 070700001
Welltech Computer Co., Ltd.
Example 2: Matched target ANI and matched DNIS together
Matched
Prefix
0701
080100001 DNIS DNIS
Matched
Target
ANI
(Matched
ANI first)
OP
Target
DNIS
Before DM
ANI: 070700001
DNIS:
0282265699
ANI: 070100001
DNIS: 080100001
Start
Position
0 15 080100001
0 15 171222
Stop
Position
Replace
Value
After DM
DNIS: 080100001
ANI: 070100001
DNIS: 171222
ANI: 070100001
Call Interception
Call Interception can provide interception service through target call
number, be sure that you have an external Recorder Server like Welltech
Rec5600.
Start Path: Service > Call Interception
Figure 5.3-1
Parameter Description:
• Target Number: The calling or called number to be recorded and monitored.
• Description: Description
Device ACL
Device ACL is used to check whether the subscriber is using the dedicated
device or not. The system is able to check the User Agent for device validation.
Start Path: Service > Device ACL
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Figure 5.4-1
Parameter Description:
• Device: Device type name
• User Agent: SIP User Agent Name
• Desc: Description
Welltech products User Agent examples:
SIPPBX 6200: __sippd
WG5250/WG5500: __tgateway
SIPIVR6800: __sipivr
Centrex6850: __sipivr
WellREC5600: __iprec
SIPPBX6200: __sippbx
WellXFER: __sipxfer
WellBG5800: __sipsbc
WG35xx: FXS_GW
LP201: SIP201
LP302: SIP Phone
RTP Resource Group
This feature can be used only for Welltech external RTP resource server.
By using this feature, SIPPBX 6200 can have more concurrent RTP proxying
channels.
Start Path: Service > RTP Resource Group
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Figure 5.5-1
Parameter Description:
• Group ID: RTP Server Group ID
• Description: Description
Click the Detail button:
Figure 5.5-2
Parameter Description:
User ID: Subscriber ID for RTP server (Device type=RTP server)
Priority: The RTP Server priority
After define the installed RTP resource server, you can set the preferred
RTP proxy server, in subscriber menu, to be used. If the preferred RTP server
is out of order, SIPPBX 6200 will use internal or other RTP resource severs.
NAT Group
NAT group can be used for enterprise user. When two subscribers have
same NAT group defined, SIPPBX 6200 will not use NAT Proxying when both
subscriber have same NAT group. You have to define NAT group definition by
detail, or same IP address detect policy will be used. It is useful when your
branch office has multiple IP address on their NAT server. By setting it, those
calls in branch office will route voice in their LAN directly.
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Start Path: Service > NAT Group
Figure 5.6-1
Detail:
Figure 5.6-2
Parameter Description:
• IP Address: public IP address on NAT
• Sub-mask: network mask
DNIS Screening Group
DNIS screen group can be used to limit the called prefix. First you will
need create the screening group and assigned it for the office as the general
screening filter.
Notes: It is valid only when Toll Restriction is disabled.
Start Path: Service > DNIS Screening Group
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Detail:
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Figure 5.7-1
Figure 5.7-2
Parameter Description:
• Screening Prefix: Called number prefix
• Screening Type: Allow or disallow
When all of detail records are set to Allow (prefix match), only on-list DNIS
can get through. When all TEL are set to disallow, only on-list DNIS will be
screened. Otherwise, “disallow” has higher priority than “allow”.
Emergency Call
To have subscriber-based emergency call setting, please define the
required emergency call here and select it on subscriber basis.
Start Path: Service > Emergency Call
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Click the Detail button:
Welltech Computer Co., Ltd.
Figure 5.8-1
Figure 5.8-2
Parameter Description:
• Emergency Number: Emergency called number (e.g. 911)
• Routed Number: Actually called number to be dial out (e.g. 0222211111)
Voice Codec Group
Voice codec group can be used to limit the supported voice codec. You
can assign the voice codec group for each subscriber.
Start Path: Service > Voice Codec Group >New
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Figure 5.9-1
Parameter Description:
• Group ID: The voice codec group ID
• Codec: The supported voice codec of this group
• Description: The description
Apply Change
When you load a new working configuration or change any configuration, you
need click Configuration > Apply Change to take effect as figure 5.10-1.
Figure 5.10-1
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Chapter 6 Enhance Service Reference
Here is the parameter setting for the PABX value added services such as auto
attendant, voice mail, extension and office etc.
System
Here come some prefix or system wide parameters.
Start Path: Enhance Service > Config > System
Figure 6.1-1
Parameter Description:
• Enhance Service Account: Service account of enhance service. The
system will automatically add the account you inputted here.
• Reserved Auto Attendant & VMS Channels: The min channels reserved for
the auto attendant and VMS
• Reserved conference Channels: Min conference channels to be used
• Reserved CRBT/Announcement Channels: Min announcement/CRBT
channels to be used
• Min Wakeup Call Channel Count: Minimum wakeup call channel will be
used (reserved) during the wake up call service.
• Max Wakeup Call Channel Count: Maximum wakeup call channel will be
used for wake up call service. Over this limit, the rest of request will be
queued.
• Server Broadcast Prefix: The prefix of the broadcast. The broadcaster uses
this prefix + broadcast-group-id to select which broadcasting group will be
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used. 6200 will initiate all calls out-to broadcasted number. It is required to
have the specified CPE device to understand the broadcasting server
request. SIPPBX 6200 can allow having up-to 16 parties of current
broadcasting target for 6200s and 30 parties for 6200. The system wide
total broadcasting target is same as DSP channel license.
• Server Broadcast Max Time: The max broadcasting time in second for a
broadcasting request. When exceeding the max broadcast time, 6200 will
disconnect the calls.
• Server Broadcast No Answer Timeout: The default maximum time (in
second) to wait the broadcasted party pick up the phone.
Summation of (Reserved Auto Attendant + Reserved VMS Retrieve +
Reserved Conference + Reserved Wakeup Call) cannot exceed the limitation
of DSP resource. If it is smaller than max DSP resource, the rest of DSP
resource will be dynamically allocated by required service.
• Reject Any Call from Enhance Service: Reject the call from Enhance
Service itself. It is recommended to be “enable”.
• Service Setting IVR Prefix: The prefix of the service setting IVR. The
Service Setting IVR allowed extension changed service without complex
service code. The user can dialing into the service setting IVR prefix and
the IVR will guide it through voice for service settings.
• Service Setting Max Operation Timeout: The max time of service setting
operation.
• Service Setting Default Language: The default language of service setting
IVR prompt
• Reserved Service Setting Channel: The reserved channels for service
setting IVR
Interface
Start Path: Enhance Service > Config > Interface
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Figure 6.2-1
Parameter Description:
• UDP Port: The local listen UDP Port
• Min DTMF Interval (msec): The minimum interval in mega second to
resend the DTMF signal
• G.711 Psize: G.711 transmission packet size (default: 20ms)
• G.723 Psize: G.723 transmission packet size (default: 30ms)
• G.729 Psize: G.729 transmission packet size (default: 20ms)
• NAT Traversal: Enable/Disable NAT Traversal feature of enhance service.
SIP
Start Path: Enhance Service > Config > SIP
Figure 6.3-1
Parameter Description:
• SDP Priority 1: SDP priority 1
• SDP Priority 2: SDP priority 2
• SDP Priority 3: SDP priority 3
• SDP Priority 4: SDP priority 4
• SDP Order: Select Local or remote SDP order to match the remote SDP for
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codec selection
• DTMF Relay: DTMF transport type selection
• Send 100: Enable to send SIP 100 Trying when AA/VMS receive a call
• Send 180 or 183: Enable to send SIP 180 or 183 when AA/VMS receive a
call
• RFC 2833 Select Order:
- Local RFC2833 Order: Use local RFC2833 payload type priority to
match
- Remote RFC2833 Order: Use received RFC2833 payload type priority
to match
• RFC 2833 Payload Type Priority 1: 96 or 101. It is recommended to use
101 and then 96.
• RFC 2833 Payload Type Priority 2: 96 or 101
Advance:
Figure 6.3-2
Parameter Description:
• UDP Port: The local UDP port on which the SIP service listened
• Retransmission T1 (ms): T1 determines several timers as defined in
RFC3261. For example, when an unreliable transport protocol is used, a
Client Invite transaction retransmits requests at an interval that start at T1
seconds and doubles after every retransmission. A Client General
transaction retransmits requests at an interval that starts at T1 and doubles
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until it reaches T2. (Default Value: 500ms) **
• Retransmission T2 (ms): Determines the maximum retransmission interval
as defined in RFC3261. For example, when an unreliable transport protocol
is used, general requests are retransmitted at an interval which starts at T1
and doubles until reaches T2. If a provisional response is received,
retransmission continue but at an interval of T2. (Default Value: 4000ms) **
• Retransmission T4 (ms): T4 represents the amount of time the network
takes to clear message between client and server transactions as defined
in RFC3261. For example, when working with an unreliable transport
protocol, T4 determines the time that UAS waits after receiving an ACK
message and before terminating the transaction. (Default Value: 5000) **
• Invite Linger Timer (ms): After sending an ACK for an INVITE final
response, a client cannot be sure that the server has received the ACK
message. The client should be able to retransmit the ACK upon receiving
retransmissions of the final response for inviteLingerTimer milliseconds.
• General Linger Timer (ms): After a server sends a final response, the
server cannot be sure that the client has received the response message.
The server should be able to retransmit the response upon receiving
retransmissions of the request for generalLingerTimer milliseconds.
• Provisional Timer (ms): The provisionalTimer is set when receiving a
provisional response on an INVITE transaction. The transaction will stop
retransmissions of the INVITE request and will wait for a final response
until the provisionTimer expires. If you set the provisionTimer to 0, no timer
is set. The INVITE transaction will wait indefinitely for the final response.
• Cancel General No Response Timer (ms): When sending a CANCEL
request on a General transaction, the User Agent waits
cancelGeneralNoResponse Timeer milliseconds before timeout termination
if there is no response for the cancelled transaction(Default Value:
10000ms).**
• Cancel Invite No Response Timer (ms): When sending a CANCEL request
on a Invite transaction, the User Agent waits
cancelInviteNoResponseTimer milliseconds before timeout termination if
there is no response for the cancelled transaction(Default Value:
10000ms).**
• General Request Timeout Timer (ms): After sending a General request, the
User Agent waits for a final response generalRequestTimeoutTimer
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milliseconds before timeout termination (in this time the User Agent
retransmits the request every T1,2*T1,…T2,…milliseconds)**
• Max Subscriptions: The maximum number of subscriptions
• Subscription Alert Timer (ms): Indicates the time in milliseconds that an
alert is given, before subscription expiration.
• Subscription Notify Timer (ms): Indicates the maximum time in milliseconds
that a subscription waits for a first NOTIFY request after receiving a 2xx
response to a SUBSCRIBE request.
• Subscription Auto Refresh: Specifies whether to send a refresh
SUBSCRIBE request when the subscription is going to be expired.
• Session Expires (sec): This is the setting of initial session timer expires
time according to rfc4028 - Session Timers in the Session Initiation
Protocol.
• Min Session Expires (sec): The minimum session timer allowed when
receiving a call with session timer value according to RFC 4028.
• Replace ANI with Valid Display Name: Enable replace calling number with
display name or not
Voice Mail
Set subscriber’s voice mail parameter here.
Start Path: Enhance Service > Config > Voice Mail
Figure 6.4-1
Parameter Description:
• Max VMS Messages K
• Min Free Space Percent (%): The minimum free space percent. If the fr
space under it, the system will start to delete those read messages for that
subscriber.
• Access Message Max OP Timeou
operation
eep Days: The max VMS messages keep days
ee
t (sec): The max time of access message
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• VMS Access Code 1: Use extension number as the VMS user to access
voice mail. No extension input is required.
• VMS Access Code 2: Will ask for extension n
umber and password
authentication to access VMS.
• Leave VMS Directly Prefix: This
is the prefix to be used for entering the
leaving message directly without prompt. The system will automatically
create the required prefix. The users only need to dial this prefix and
extension number for leaving the message. It is useful for operation
console to transfer the customer into a voice mail.
• Leave Message Max OP Timeout (sec): The max time
of leaving message
operation
• Response
Busy When Come From Conference: Enable the response busy
when the call comes from conference or not. It is recommended to set to
enable.
Click the Notify button to set voice mail notice parameters as figure 6.4-2:
Figure 6.4-2
Parameter Description:
• Enable Notify: Enable e
mail notice or not
• SMTP Mail Server: SMTP Mail Server
• Auth Mode: Authentication mode, none
, cram MD5,login,plain or NTLM
• SMTP Reconnect Interval (sec): SMTP reconnect interval time
• Send Mail Max Retry: The max retry number of sending mail
• Query VMS DB Interval (sec): Query VMS DB interval time (it
greater than 120 seconds)
• From Address Automatic Ass
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should be
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not. If it is set to automatic, the email From address will be
Telno@SIPPBX_IP.
• Manual Assign Address: Manual assign the FROM mail address
• Email Priority: Email priority: none, low, normal or high
• Email Subject: Email subject. The following variable can be used for VMS
Email notice:
- $ANI$: Calling Number
- $MSG_TIME$: The voice mail message date and time
- $MSG_FILE$: The voice mail message file name
• Char set: Character set setting
• MP3 Attach: attach MP3 file or not
• Rich Email: Rich email(with logo and logo file) or not
• Rich Email Logo: Add rich email Logo or not
• Logo File (GIF only): Logo file name
• MP3 Attached file bit rate: The bit rate to attach the MP3 file. You can adjust
this rate to have better performance. It is recommended to use 24K or 32K
sample rate. The higher bit-rate will have bigger MP3 attachment mail size.
Click the MWI button to set message waiting indicator parameters as figure
6.4-3:
Figure 6.4-3
Parameter Description:
• MWI Service Account: Message Waiting Indicator Service code. You need
to have this service code setup in your CPE in order to get the voice mail
display message.
• MWI Listen UDP Port: MWI listen UDP Port (8080-8085)
• Authentication Mode: Reply 200 or 202 when receive a subscription.
• Min Expire Time (min): Min expire time of MWI subscription in minutes.
• Max Subscriptions: Max subscriptions for the MWI service. Please don’t
change it unless under Welltech instruction.
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Auto Attend
Start Path: Enhance Service > Config > Auto Attend
Figure 6.5-1
Parameter Description:
Call Transfer Timers:
• AA Max Operation Timeout (sec): The max time for AA operation
• Operator Retry Interval (sec): Refer operator interval time
• System No Answer Timeout: The max time to wait the system to answer
the call
Click the Transfer button:
Transfer condition is used map Auto Attendant reason to SIP response code.
For example, you can define 486 and 480 to be processed as busy.
Figure 6.5-2
Parameter Description:
• Reason Code: SIP response code
• Condition: Auto Attendant transfer handling code
Conference
Conference parameters can be defined here.
Start Path: Enhance Service > Config > Conference
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Figure 6.6-1
Parameter Description:
• Conference Prefix: Conference Service URI
• Virtual Conference Prefix: Virtual conference prefix code
• First Response Timeout (sec): The maximum time to wait new incoming
call after send 302 out to CPE
• No Answer Timeout (sec): The maximum time to wait the remote party
answer
• Max Participants per Room: The max participants x per room
• Tone Notification: Enable tone notification or not when new participant is
joined or leaved.
• Close Conference for Creator Leave: Close the conference or not when
creator leave
• Max Simultaneous Speaker: The max simultaneous speaker number
• Conference Prompt Default Language: The default language of conference
prompt
• Max time for one participant (min): The maximum time to wait the second
participant to join the conference. When it exceeds the max waiting time,
conference will stop the conference in order to get better DSP channel
utilization.
Announce Service
Announcement service is used for company wide coloring ring back tone or
non-VMS users’ announcement.
Start Path: Enhance Service > Config > Announce Service
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Figure 6.7-1
Parameter Description:
• Announce Timeout (sec): The maximum time to execute for play
Announcement
• Announce Prompt Default Language: The default language of announce
prompt
• CRBT Timeout (sec): The maximum time to execute for play CRBT
• CRBT Default Language: The default hold tone files to be used for call
transfer.
Language
Start Path: Enhance Service > Config > Language
The system can support default Chinese, English and Japanese language.
You can have local language together with text to speech file. For those texts
to speech (e.g. date, time or message count), you have to prepare your TTS
files as follows:
1. Download the TTS00.zip
2. Re-record all files extracted in G.711 Mu-law 8K raw file format.
3. ZIP the file name as the language ID (e.g. Language ID: 6)TTS06.zip
4. Use upgrade to put the TTS06 to 6200.
For those voice file (not TTS part), you need to upload the required voice files
into corresponding Language ID directory by using voice file manager. The file
format is WAV file format based on G.711 Mu-law 8K mono.
Figure 6.8-1
Parameter Description:
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• Language ID: Language ID
• Language: Language
• TTS: Enable the TTS or not. When you enable the TTS, the 6200 will try to
load corresponding TTS0?.zip. If it is failed, 6200 will use English for TTS.
Call Flow Editor please refer to Chapter 7 Call Flow Editor
Wakeup Call
The SIPPBX 6200 can call the extension to wakeup the user. Please refer
Appendix 7 Step by Step example for Wakeup Call
Here you can set the wakeup call parameters.
Start Path: Enhance Service > Config > Wakeup Call
Figure 6.9-1
Parameter Description:
• Wakeup Call Prefix: Wakeup call service prefix code
• Max Tolerance Time (mins): Used to indicate the maximum tolerance time
for SIPPBX 6200 to call the extension to wakeup the user, when the time
exceeds the defined time here, the SIPPBX 6200 will not call the extension.
• Max Retry Count: The max retry times when the extension which will be
waked is busy
• Retry Interval (secs): The interval for SIPPBX 6200 to retry the extension
when it is unreachable.
• Wakeup Call Prompt Language: The default prompt language for the user
listened
• Max Operation Timeout (secs): The maximum operation time in seconds.
The SIPPBX 6200 will disconnect the wakeup call automatically if the
extension is not hung up and when the Wakeup Greeting Report Count is
set to Repeat Forever.
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• No Answer Timeout (secs): The maximum ringing time when no person
answers the wakeup call
• Wakeup Greeting Repeat Count: The wakeup greeting broadcast times
when the extension is not hung up. If you set the count is 3, then the
SIPPBX 6200 will announce the wakeup greeting 3 times and disconnect
the operation automatically.
- Repeat Forever: The SIPPBX 6200 will announce the wakeup greeting
until the called party hang up the call or reach the max operation time.
FAX Setting
The SIPPBX 6200 can support fax service to a common account or personal
fax (license is required). User can retrieve the FAX through WEB. And also
the system can be set to send a mail with the attached fax PDF when a new
fax is received. Here you can set the FAX parameters.
Start Path: Enhance Service > Config > FAX Setting
Figure 6.10-1
Parameter Description:
• Max FAX Messages Keep Days: The maximum FAX messages keep days
• Min Free Space Percent (%): The minimum free space percentage is
required to be kept by the system. If the free space is under it, the system
will start to delete those read messages for that subscriber. For those
un-read message will still kept it.
• FAX Prefix: The prefix of the FAX Service. The Extension can receive the
FAX service by following ways.
3. FAX in through Auto Attendant: The Caller dials into Auto Attend first
then transfer to “FAX Prefix+Extension” and press start fax of the fax
machine.
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4. Direct Fax in number: Create a new extension as the fax DID number
and enable unconditional forward to “FAX Prefix+Extension”.
The Caller can send fax by dialing this fax number directly.
The received FAX messages will be stored as PDF format.
The fax message can be retrieved through web interface or send a email
to the user with the fax PDF attachment.
• Send T.38 Re-invite Delay (sec): The delay to send re-invite for T.38 after
call connected.
• T.38 Re-invite Accept Timer(sec): The timer for waiting 200 OK after the
T.38 re-invite. When timeout is happened, the call will be disconnect.
• Use ECM: Enable or disable the Error Correction Mode for FAX service
• UDPTL EC Entry Count: The T.38 redundant depth. The default value is 2.
• Max Operation Timeout(sec): The maximum operation time in seconds.
The SIPPBX 6200 will disconnect the FAX call automatically if the FAX IN
time over than it.
Device Auto Configuration
The SIPPBX 6200 provided the Extension Device Auto Configuration feature.
System Administrator can easy managing the extension device without setup.
Most of setting can be automatic provisioned to CPE (current support LP600N
only).
Here you can set the Extension Auto Configuration parameters.
The following parameters will be provisioned to
- SIPPBX discovery (when using VLAN and DHCP)
- Telephone number and PBX related setting
- Access code required for LP600N
- Office and personal phone book
- Office dialing plan
- Common Device Configuration
- Firmware Upgrade
The most settings were full integrated into SIPPBX configuration. Here is for
the device (LP600N)’s common configuration and firmware upgrade.
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Start Path: Enhance Service > Config > Device Auto Configuration
Figure 6.10-1
Device Cfg:
This is the place to upload the device configuration setting file. Please get
the CPE template from Welltech and modify when necessary. The current
model supported by SIPPBX 6200 is LP600N and the configuration file need to
be lp600n.cfg. In the lp600n.cfg, there is a parameter to indicate the firmware
name. When the firmware name was different from the existing LP600N’s. The
LP600N will automatically upgrade it.
Firmware File:
Here is the place to upload the firmware file for LP600N. Once you setup a
new firmware in lp600n.cfg, you need to upload the corresponding firmware to
SIPPBX 6200 for LP600N to upgrade.
Office Profile
SIPPBX 6200 can support headquarter and multiple branch offices, each office
can have different setting including incoming number, language and call flow.
Start Path: Enhance Service > Office Profile
Figure 6.10-1
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Modify:
Click the office profile you want to modify:
Figure 6.10-2
Parameter Description:
• Office ID: The branch office ID
• Name: The branch office name
• Time Zone: The branch office time zone
- Standard: Use a predefined standard time zone (Refer to section
“Timezone to Country Mapping List” )
- Customize: Use a user defined time zone
• Auto Daylight Saving: Auto adjust daylight saving time or not
• Office Phone Number: The incoming telephone number of the branch office
• AA Primary Language : The auto attendant preferred language
• AA Secondary language: The auto attendant second language
• Broadcast Target Group: The default broadcasting group. If this branch
office’s person starts the broadcasting without specify the group ID, this
default group ID will be used.
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• Toll Restriction Group: The Office Level Toll Restriction Group. System will
search the Office Level first and then Extension Level.
• Description: The description
User Group Information
• DM Group ID: Group-wide digit manipulation applied
• SMTP Host: SMTP server host (i.e. mail.welltech.com.tw) for delivering
missed call message
• Miss Call Subject: Missed call notify subject
You can have the following variables for notify subject in order to give better
mail subject.
$FROM$: caller party number
$TO$: Called party number
$UTCTIME$: UTC Time
$LTIME$: Local Time
$DOMAIN$: SIP Domain
$HOSTIP$: Host IP address
For example:
You have a missed call from $FROM$ at $LTIME$
• Enable Presence: Whether to support SIP presence server for this office
user or not. It requires setup a subscriber for the presence server.
• Call Park: Enable call park for this office user or not.
- Call Park Location: The Call Park Location starting code (e.g. 800, and
the system will automatically add to 809, 10 locations in all.) It cannot be
conflict with subscriber or prefix.
Extension Provision
• Phone Book Refresh Time (mins): How long the LP600N will try to refresh
the phone book from SIPPBX 6200. The minimum value allowed is 30
minutes.
• Device Config Refresh Time (mins): How long the LP600N will try to get the
lp600n.cfg for parameters and firmware upgrade. The minimum value
allowed is 30 minutes.
• Firmware Update Time: When LP600N detect a firmware upgrade is
required. This is the time it will do the restart to complete the upgrade. If
you set the time to “Update A.S.A.P”, LP600N will to the restart after it
download the new firmware and there is no call is talking.
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Service Code
Service code definition for the user group
Figure 3.3-2
Parameter Description:
• Service Code: Telephony Keypad used for the service code
• Service Type: Applied service type
- Call Return: Call Return service, only Access code(e.g.*600)
- Call Park: Call Park service, Access code + ext.(e.g. *800+301)
- Camp On: Camp Call service, Access code + called user ID (e.g.
*500+070700001)
- Call Pickup: Call Pickup service, Access code+ ext. (e.g.*700+301)
The others please refer to the examples below:
Forward Service:
Service
Enable unconditional forward
Enable no answer forward
Enable busy forward
Access
Parameter (optional) Example
Code
*2010933111666
*201 Forward number
*201 (use existing setting)
*2020282265699
*202 Forward number
*202 (use existing setting)
*2030936111222
*203 Forward number
*203(use existing setting)
Enable unavailable forward
Enable don’t disturb
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*204 Forward number
Don’t disturb time 1
*205
(hhmmhhmm)
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*204302
*204(use existing setting)
*20518002359
Welltech Computer Co., Ltd.
Don’t disturb time 1 & 2
(hhmmhhmmhhmmhhmm)
*205(use existing setting)
*2051800235901000900
Enable Notify
Enable Fine Me
Enable CRBT
Enable Announcement
Enable VMS
Disable unconditional forward
Disable no answer forward
Disable busy forward
Disable unavailable forward
Disable don’t disturb
Disable Notify
Disable Fine Me
Disable CRBT
Disable Announcement
Disable VMS
*206 n/a *206 (need pre-config by web)
*207 n/a *207 (need pre-config by web)
*208 n/a
*209 n/a *209 (need pre-config by web)
*210 n/a
*301 n/a *301
*302 n/a *302
*303 n/a *303
*304 n/a *304
*305 n/a *305
*306 n/a *306
*307 n/a *307
*308 n/a *308
*309 n/a *309
*310 n/a *310
*208 (need pre-config by web)
*210 (need pre-config by web)
Hide ANI Service:
Service
Hide ANI
Show ANI
Access
Parameter Example
Code
*314070100001
*5 Dialed number
(Hide caller ID)
*21407010062
*214 Dialed number
(Show caller ID)
Pickup Call Service:
Service
Global Pickup *0 n/a *0 (global pickup)
Group Pickup *1 n/a *1 (group pickup)
Call Pickup *213 n/a *213 (call pickup)
Camp on *211 n/a *211 (camp on)
Access
Parameter Call Pickup
Code
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Call return *212 n/a *212 (call return)
VAD Service:
Service
Enable Call Waiting
Enable Privilege Access
(Valid only when Toll Restriction
is disable)
Enable Caller ID
Disable Call Waiting
Disable Privilege Access
Disable Caller ID
Access
Parameter Example
Code
*215 n/a *215
*216
070700001
(user’s password)
*216 (enable privilege access)
*217 n/a *217
*315 n/a *315
*316 n/a
*316 (disable privilege access)
*317 n/a *317
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Pickup
Grouping the subscribers for group pickup service, you can set a
subscriber to belong to a pickup group.
Figure 3.3-3
Parameter Description:
• Pickup Group ID: pickup group ID
• Description: Description
Holiday
You can define each office’s holiday settings.
Click the Holiday button to set the holiday of the branch offices themselves.
Figure 6.10-3
Parameter Description:
• Date: Holiday date, the format is MM/DD.
• Description: The description
Work Time
Click the Work Time button to define the office’s working time here for auto
attendant call flow.
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Figure 6.10-4
Parameter Description:
• Holiday: Non-working weekday
• Office Hour: The working time setting for a week
• Lunch Break: The lunch time setting for a week
• SUN ~ SAT: The week day
Phone Book (Office shared phone book)
Click the Phone Book button to management the office’s shared phone book.
This phone book will be synced to LP600N through auto configuration feature.
Figure 6.10-5
Parameter Description:
• Contact Name: User name. (English only)
• TEL No: User phone number
Dial Plan
Click the Dial Plan button to define the office’s dial plan. The dialing plan will be
synced to LP600N through auto configuration feature.
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Figure 6.10-6
Parameter Description:
• Dialed Prefix: The prefix matched (the longest prefix will be matched first)
• Max Digits: The max digit lengths for the matched dialed prefix
Example:
Condition: Dialed Prefix set to 09 and max digits set to 10
Result: When users dialed 09xxxxxxxx, LP600N will start to make call
immediately without waiting for stop key or inter digit timeout.
Department Profile
Each extension can be assigned to a department for account purpose (CDR).
Thus you can query CDR and summary it by different department. This is the
place to define the department profile.
Start Path: Enhance Service > Department Profile
Figure 6.11-1
Parameter Description:
• Department ID: The department ID
• Name: The department name
• Description: The description
Extension
The extension is representing a user which normally has a physical device such
as IP phone or a line of VoIP gateway etc.
Start Path: Enhance Service > Extension
Figure 6.12-1
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Modify
Click the extension you want to modify:
Figure 6.12-2
Parameter Description:
• Active Mode: The subscriber user is active or inactive
• TEL No/Account: Register TEL no or user account
• User Password: Register user password (device password only)
• Web/VMS Password: Password used for web access or retrieve VMS
• Office: Belonged office ID
• Authentication Mode: Authenticate subscriber by MD or not
- None: None
- Register Only: Authenticate subscriber only for register
- Register Invite: Authenticate subscriber for register and each call
• Department: Belonged department ID
• Toll Restriction Group: The Extension Level Toll Restriction Group. System will
search the Office Level first and then Extension Level.
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• DNIS Screening Group: DNIS screening group
• Call Authorization Mode: Send authorization to Radius server or not
- None: None
- Radius: Send to RADIUS for call permission
• Emergency Group: Emergency call group
• Caller ID Mode: Displace caller ID or not
- Inhibit: Hide the called party number
- Transparent: Pass through the caller ID
• Device Type: Subscriber device type
- Subscriber: Subscriber user
- Softphone: It is Welltech soft-phone device type. Welltech SP-365 is
required as the soft-phone.
• Hunting Method: Call forking method
- Sequential: Call hunting each contact in sequence
- Parallel (answer): Send multiple call invites to multiple contacts
simultaneously. When a user pickup the phone, disconnect others
contacts.
• Preferred RTP Group: Preferred RTP resource server group to be used
• Register Type: Subscriber register type
- Dynamic: Subscriber need send register message for availability
- Predefine: Subscriber will be handle as a permanent user
Predefine URI1: Predefine subscriber URI1 (i.e.
sip:9001@203.69.216.53)
Predefine URI2: Predefine subscriber URI2 (i.e.
sip:9001@203.69.216.54)
- Predefine/NAT: Subscriber will be handled as a permanent NAT user
(manual IP/Port mapping is required)
Predefine URI 1: Predefine NAT subscriber URI (i.e.
sip:8001@192.168.19.61: 7777)
Public TA: mapped NAT Server IP address and port (i.e.
61.218.42.217:5060)
- LCS Dynamic: This is used only when the subscriber type is a regular
subscriber and also a LCS user. You can use IP phone or CPE to register
this account and also it will ring the LCS communicator client software
simultaneously.
• RTP Proxy: Use RTP Proxy or not
- Yes: Always use the RTP Proxying
- No: Always not use RTP Proxying
- Auto: Automatic decide to use RTP Proxying or not (recommended)
- Recorder: The subscriber is required voice recording
- Recording on Demand: Use Recorder on demand service (WellRec 5600 is
required)
- Auto (n-NAT): This is only used when you have a NAT box behind a NAT
box. It is mainly for testing or a special environment.
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• NAT Group: NAT group can be used for enterprise user. When two subscribers
have same NAT group defined, SIPPBX 6200 will not use NAT Proxying when
both subscriber have same NAT group.
• Max Register Time: The maximum register time when a user is coming from
public network
• Max NAT Register Time: Time: The maximum register time when a user is
sited behind NAT
• First Response Time: The maximum time to wait for response. It’s depended
on the network speed.
• No Answer Timer: The maximum time (in second) to wait the remote party
answer (pick up phone).
• Max Contact Allowed: The maximum contact allowed for a subscriber. The
new contact will not able to register when old one doesn't free up.
• Pickup Group: Pickup group for per subscriber
• Device_1, 2: Limited device type for the subscriber
• Max Concurrent Call: The maximum of concurrent call
• Call Validation: The call validation type: none, update or invite
• Over Max Contact Rule: Over Max Contact Rule, reject or update. Or use
Global Setting (Configuration > System). When set to reject, the system will
reject the new register. Otherwise, it will drop the current contact and use the
new one.
• AAA Sending Stage: Send AAA message before or after DM or use Global
Setting (System > Advance)
• Codec Group: Belonged codec group. When you select a codec group, only
selected voice codec can be get through. The others will be filtered. It is very
useful when you want to limit your users’ codec.
• Dedicate Outgoing UAC: When enable it, the subscriber will use the selected
UAC for making outgoing call. You can use PSTN number together to provide
a DID and DOD feature. Assign a special UAC device for outgoing calls
• MAC Address 1,2,3: LP600N’s MAC address which is used for plug & play &
auto configuration. There is no setting required for LP600N to run. Up-to 3
MAC addresses can be set for each subscriber. However, each MAC can be
assigned to a subscriber only.
• Effective Period: The subscriber effective period (Format:
yyyymmdd-yyyymmdd)
• Remove Tag for Cancel: When cancel the call, remove the “to tag” (for CISCO
device only)
• Disallow Register From NAT: Enable this option will not allow a subscriber to
register behind NAT. In other words, this subscriber will never consume the
RTP resource.
Enhance
• Virtual Conference Active Mode: To enable the virtual conference service or
not.
• Virtual Conference Room: Virtual conference room number. The final virtual
conference room number is prefix + inputted number.
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• Virtual Conference Join PIN : When enter the conference join PIN, the
conference participant requires the password checking in order to join the
conference room. Also when the conference owner want to active the virtual
conference from other extension, the system will request to input create PIN
for activation ( the creator Pin is same as VMS password)
• Transit Call: Allow to call to PSTN number inside the Auto Attendant or not.
The administrator needs to add transit call component in auto attendant call
flow editor. The sample call flow can be downloaded from Welltech support
web site.
• Global Broadcaster: Does the user can broadcast any broadcasting group or
not. It just likes a super-user of the broadcasting service.
• VMS Access Type: Allow to use WEB or Voice to access voice mail or not.
• Max Keeping Voice Message: The maximum keep days of voice message
• Default VMS Language: The default VMS language
• VMS Personal Greeting: Enable personal greeting or not
• VMS/FAX Notice: Enable Email notice for VMS/FAX message or not
• FAX Service: Enable or disable the FAX-IN service. When enable it, the
extension will have the ability to receive its own fax.
• Notice Email Address: Email address for the VMS service
Information
• User Name: The user contact name
• Contact TEL: The telephone number
• Email Address: The email address of the subscriber
• Address: The contact address
• Description: Description
Service:
Click the Extension > service to modify telephony service for the selected user.
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Figure 6.12-3
Parameter Description:
Forward Service:
• Forward Subscriber Only: Forward to PBX user only. When it is enabled, the
forward target will only allow extension only, not PSTN.
• No Answer Forward: Forward to the URI when the subscriber has no answer.
• Unconditional: When enabled, any calls to this subscriber will be forward to
this URI unconditionally. You can use SIP URI or subscriber ID here.
• Busy Forward: Forward to the URI when the subscriber is busy. You can use
SIP URI or subscriber ID here. You can use SIP URI or subscriber ID here
• Unavailable Forward: Forward to the URI when the subscriber is unavailable
(not registered). You can use SIP URI or subscriber ID here. You can use SIP
URI or subscriber ID here
• 181 for Call Forward: Send 181 for Call Forward
• Announcement before Forward: The system will play announcement before
forward the call to new number.
• Find Me: Locate subscriber based on different time segment when the original
(registered) contact cannot be reached.
- Find Me Hunting First: Hunting find me contact first.
- Hunting Subscriber: Applicable only for Find Me hunting checked will hunt
subscriber registered contact when find me can’t be reached.
• Number Change: Change the original number to a new number. When enable
this feature, the system will play announcement for the changed number.
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• Auto Call Forward: Auto call forward to the changed number or not after
number changed.
Call Pickup:
• Group Pickup (Picked): Allowed to be picked-up within group or not
• Global Pickup (Picked): Allowed to be picked-up globally or not
Personal Screening Service:
• Personal ANI Screening: Personal ANI screening can be used to filter the
caller based on caller ID. For all TEL are set to allow (full match), only on-list
ANI can get through. For all TEL are set to disallow, only on-list ANI will be
screened. Otherwise, “disallow” has higher priority than “allow”.
• Personal DNIS Screening: Personal DNIS screening can be used to limit the
called prefix. For all TEL are set to allow (prefix match), only on-list DNIS can
get through. For all TEL are set to disallow, only on-list DNIS will be screened.
Otherwise, “disallow” has higher priority than “allow”
• Do Not Disturb: Up-to 2 time segments can be set to reject all incoming calls.
VAD Service (Value Added Service)
• Coloring Ring Back Tone: Coloring Ring Back Tone service
• VMS: Voice Mail Service
• Server Hold Tone: Enable Server Hold Tone or not. When you enable it, the
SIPPBX will play the hold tone music instead of CPE device play the hold
tone.
• Announcement Service: Enable the announcement service or not. Normally,
you don’t need enable it since in office will enable voice mail service.
Subscriber Service
• ANI Replacement: Replace calling number for Gateway or all subscriber
- Replace ANI: Replace calling number. It is used to integrate PSTN number
into VOIP number. Thus the receiver will get the corresponding PSTN
number for call back. To use this function, normally need PSTN provider’s
permission to send the caller ID.
- Replace Type: Whether to replace the calling number when send the call to
gateway only or not.
- PSTN Number: This number will be handled as a PSTN number. It will
work like you have a second number in SIPPBX 6200. 6200 will look at the
PSTN number first.
• Short Code: short code to be used within same group
• Display Name: Change SIP display name to the set name.
• Disable call forward display name: Add display name as subscriber id in SIP
from header when call forward caller mode is "forwarder".
• Support Video: Enable to support video or not. If not, video SDP will be set to
disable.
• Missed Call: Missed call notify service
• Disable Call Waiting: Disable call waiting feature. When disable call waiting
features, the second incoming call to the user will be rejected by 6200.
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• Disable Conference Call: Disallow to call a conference call
• Server Transfer: The server will do the transfer instead of send to CPE. It is
recommended to use it only when CPE doesn’t support call transfer features.
It is only happened when the user is transferred party.
Security
• Disable Un-Register All: Disable Un-Register all
• Disable RADIUS Billing Send: Disable RADIUS Billing Send
• Reject Anonymous Call: Reject the anonymous incoming call or not
Misc
• Sync to Address: Set SIP to header.
• Response to Sending Port: Response to CPE sending port instead of public
contact port.
• Response to Top Via: Response SIP message to the top via IP address and
port.
• CTI: Computer Telephony Integration (reserved item) Internal use only
• Auto Subscriber for MWI: Automatically subscriber MWI service to MWI server
when the user is registered.
• Disable Qop: Disable sending qop tag in SIP 401 and 407 authentication
header.
• Auto Subscriber for Presence: Automatically subscriber Presence service to
Presence Server when the user is registered.
• Register Event Log: The system will trigger a system event when the
subscriber is registered and unregistered. For a predefine user, the system will
also fire a system event when the predefined user is no response and
responding.
• Secretary: The secretary service. All the calls dial to the leader must be
answered by the secretary. And then the secretary will judge whether the call
should be transfer to the leader. This will avoid disturbing unnecessarily.
Parameter Button
• Copy: Copy service setting from a subscriber
• Mask: Set the subscriber visual view of the service. If you uncheck the mask of
a service, the subscriber login will not able to see it.
Wakeup:
Click the Extension > Wakeup to modify one time/countdown wakeup call for the
selected user.Select the wakeup call type and parameters for the extension here.
Figure 6.12-4
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Parameter Description:
• One Time Wakeup Call: The wakeup call only work one time. For example:
you set a wakeup call to 8:00am in the 10:00am, and then the wakeup call will
ring tomorrow 8:00am.
• Wakeup Time (hh:mm): Set the wakeup time here. The format is hh:mm.
• Countdown Wakeup Call: The countdown wakeup call
• Countdown Minutes: Set the countdown wakeup call minutes here
Phone Book: (Extension Level)
Click the Extension >Phone Book button to management the extension’s
personal phone book for Extension Auto Configuration purpose.
Figure 6.12-5
Parameter Description:
• Contact Name: User name. (English only)
• TEL No: User phone number
Import: (Extension Data/MAC Address)
Click the Import button to import the extension’s profile data based a exported file
or a dedicated LP600 extension profile based on MAC address when using
Extension Auto Configuration feature. When using MAC address for LP600, only
a few frequently used fields are required. The others will be auto provisioned.
Figure 6.12-6
Parameter Description:
• Import Type: The type of import
Extension Data: The extension’s profile (e.g. SIPPD_UserM.txt). It is
recommended to use an exported file as a base for modification and
import.
MAC Address: The extension data for Extension Auto Configuration which
support LP600 only. (e.g. MACTEL.xls)
• Import File: The import data file (.txt/.csv/.xls supported)(Reference Appendix
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2 for File format)
•
Export: (Extension Data/MAC Address)
Click the Export button to export the extension’s profile data or MAC Address
data for Extension Auto Configuration backup purpose.
Figure 6.12-7
Parameter Description:
• Export Type: The type of import
Extension Data: The extension’s profile. (e.g. SIPPD_UserM.txt)
MAC Address: The data for Extension Auto Configuration (LP600 only)
(e.g. MACTEL.xls)
• User ID: The range of extension to be exported. (Keep both empty for
selecting all)
• Office ID: The Office ID will be exported.
• Device Type: The Device Type will be exported.
After the system generate the required file, you will able to get the export data file
in zip format. Download and save it. (Reference Appendix 2 for File format)
Gateway
The PSTN gateway is used to setup the FXO, E1/T1 or SIP Gateway in order to
call out to PSTN world.
Start Path: Enhance Service > Gateway
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Figure 6.13-1
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Modify
Click the Gateway you want to modify:
Figure 6.13-2
Parameter Description:
• Web Password: Password used only for web access only
• Device Type: Gateway device type
- Gateway: Gateway (e.g. trunk gateway or FXO gateway)
- Gateway/RTP: Welltech WG5250/WG5200
- Proxy/RTP: Welltech SIPPBX 6200
- SIP Proxy: SIP proxy server
- IVR/VMS: IP IVR or VMS server
- IVR/VMS/RTP: Welltech IVR or VMS server
- Recorder: Welltech Recorder
- Outbound Caller: Outbound Caller
- Register UAC: Register user agent client
User Agent ID: User agnet ID in UAC
Caller Info: Display calling parting information
▪ Caller TEL No: Display original caller ID
▪ Registered TEL No: Registered UAC user ID
▪ Caller Display Name: SIP display name for original caller
- LCS Server: Microsoft LCS Server
- MWI Server: SIP Message Waiting Indicator Server
- Presence Server: Presence Server
- RTP Server: RTP Server
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- VMS (Diversion): The voice mail server which support diversion header as
draft-levy-sip-diversion-08.txt.
- Web Caller: It is used for Welltech Web call service. Welltech will provide
OCX and sample code for integrating into the customer’s Web server (Web
call license is required). It can be used for click to call or web 800 calls.
- Exchange Fax: Allow to provide the fax feature for Exchange 2007.
Regular T.38 device can work with Exchange 2007 UMS when using this
exchange fax user (Additional license is required).
- Diversion/VMS/RTP: This device type is used only for Welltech IPcentrx
6850. It will be able to greatly reduce the usage of RTP resource.
- Gateway/RTP/Fax: This device type is dedicated for Wellgate 5260.
Service:
Click the Gateway > service to modify telephony service for the selected
gateway.
Figure 6.13-2
Parameter Description:
Personal Screening Service:
• Personal ANI Screening: Personal ANI screening can be used to filter the
caller based on caller ID. For all TEL are set to allow (full match), only on-list
ANI can get through. For all TEL are set to disallow, only on-list ANI will be
screened. Otherwise, “disallow” has higher priority than “allow”.
• Personal DNIS Screening: Personal DNIS screening can be used to limit the
called prefix. For all TEL are set to allow (prefix match), only on-list DNIS can
get through. For all TEL are set to disallow, only on-list DNIS will be screened.
Otherwise, “disallow” has higher priority than “allow”
Subscriber Service
• ANI Replacement: Replace calling number for Gateway or all subscriber
- Replace ANI: Replace calling number. It is used to integrate PSTN number
into VOIP number. Thus the receiver will get the corresponding PSTN
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number for call back. To use this function, normally need PSTN provider’s
permission to send the caller ID.
- Replace Type: Whether to replace the calling number when send the call to
gateway only or not.
- PSTN Number: This number will be handled as a PSTN number. It will
work like you have a second number in SIPPBX 6200. 6200 will look at the
PSTN number first.
• Display Name: Change SIP display name to the set name.
• Server Transfer: The server will do the transfer instead of send to CPE. It is
recommended to use it only when CPE doesn’t support call transfer features.
It is only happened when the user is transferred party.
Security
• Disable Un-Register All: Disable Un-Register all
• Disable RADIUS Billing Send: Disable RADIUS Billing Send
• Trust Network: Show the trunk the real caller number. When the user uses the
anonymous method, if the SIPPBX 6200 is enable the Trust Network, the trunk
will know the real caller but will hide the real caller number for the called part.
This will enhance the security.
Misc
• Sync to Address: Set SIP to header.
• Response to Sending Port: Response to CPE sending port instead of public
contact port.
• Response to Top Via: Response SIP message to the top via IP address and
port.
• CTI: Computer Telephony Integration (reserved item) Internal use only
• Disable Qop: Disable sending qop tag in SIP 401 and 407 authentication
header.
• Register Event Log: The system will trigger a system event when the
subscriber is registered and unregistered. For a predefine user, the system will
also fire a system event when the predefined user is no response and
responding.
IXC Service (Internet Exchange Center Service)
Incoming Call
• Matched Prefix: The SIPPBX 6200 will ask to add this prefix for every call
dialed to the trunk.
• Disable Remove Prefix: the SIPPBX 6200 will not allow removing the added
prefix
Outgoing Call
• Outgoing Added Prefix: The SIPPBX 6200 will add this prefix for all calls dialed
out from the trunk
Parameter Button
• Copy: Copy service setting from a subscriber
• Mask: Set the subscriber visual view of the service. If you uncheck the mask of
SIPPBX 6200S/GS Release 3.1.0 User Guide
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