VIP-2100 is a cost effective solution for VoIP trunk gateway supporting one-,
port T1/E1 VoIP trunks that provides voice and fax over IP network. It
supports ITU-T H.323 V3, SIP RFC 2543/3261, SNMP V2, Call Detail Record,
WEB management and other useful functions to meet customer requirements.
The built-in enhanced IVR (Interactive Voice Response) and Billing Service
of VIP-2100 is suitable for prepaid and postpaid service. It can rapidly provide
value added service for customers.
- Support silent suppression for G.729A, G.723, G.729
- VAD (Voice Activity Detection)
- CNG (Comfort Noise Generation)
DTMF Transmission
- Transparent
- H.245 signal/alphanumeric
- H.323 Q.931
- RFC 2833
- SIP INFO
FAX Support
- Automatic voice/fax detection
- T.38 fax relay based on H.323 Annex D
- SIP T.38 fax relay
- Up to G3 fax
- ECM support
- Redundant T.38 packet (0-2)
- CISCO compatible
Built-in IVR & call-flow controller
- Web-based GUI Drag and Drop interface
- Full control of call behavior (one-stage or two-stage dialing)
- IVR functions
- Support time duration play back (Chinese & English)
- Power call information branch
- Collected information validation
- Active disconnect & reconnect without hang up
VIP-2100 User’s manual - 2 -
- Selected disconnect cause code & behavior
Management Feature
- OS and program upgradeable
- Console port: RS-232 port
- TELNET
- Full Web management interface & real time monitor
- Front panel LCD
- SNMP v2 (H.341) and SNTP v4 support
- User account management
- Time zone and day light saving support
- Support fixed IP and DHCP
- Support DNS and dynamic DNS
LED indicators for system status
- Power/Storage access indicators
lines
- Front panel LCD (2
x 16) status display
Power
- 90~240V auto switch
Environmental
- Operation temp: 0° C to 60° C
- Relative humidity: 5% to 95%
Dimension
- 483mm (L) x 450 mm (W) x 44mm(H)
Certification
- CE, FCC, EMI
VIP-2100 User’s manual - 3 -
10/100MB Ethernet ports *2 (host &
VIP-2100 Detail Specifications
Feature VIP-2100
Physical Dimension
1 Width
2 Height
3 Depth
4 Industrial rack mount
5 Color
6 Weight
Power / Environmental
1 Power
2 Operating temperature
3 Relative humidity
Processors & Storage
1 DSP vendor
2 Operation System
3 RAM
4 Program/Data Storage
5 OS Upgradeable
7 Program Upgradeable
Front Panel Display
1 LED status
2 LCD status
LAN Interface
483mm
44mm
450mm
Yes
Black
8Kg
90-240V auto switch
0~60 C
5%~95%
Intel Pentium, AudioCodes DSP
XP Embedded
512 MB
256 MB DOM
Yes
Yes
Power/DOM/System
Yes
1 10/100 Base Ethernet
2 IP Address Required
PSTN Interface
Customizable E1/T1 CAS
1
E1 CAS DTMF
2
E1 CAS R2 MF
3
E1 ISDN PRI Support
4
5 E1/T1 Interface Selectable
PCM law Support
6
RTP)
Yes
Loop Start FXO Hot-Line
Argentina, Bolivia, Brazil, Chile,
China, Czech-Republic, Egypt,
India, Indonesia, Israel, ITU, Korea,
Malaysia, Mexico, Philippines,
Thailand, Uruguay, Venezuela,
RomTelcom
Euro, Australia, Hong Kong, Korea,
New Zealand, QSIC
Alaw/Mulaw selectable
2
VIP-2100 User’s manual - 4 -
E&M Bell Core Feature Group D,
Wink Start, E&M Delay Start, E&M
Feature Group A Immediate Start,
E&M Feature Group B Wink Start,
E&M Feature Group D Wink
Gain Control Yes
5 Improved Echo Tail Suppression
6 Silence Suppression
VAD
7
Maintenance
1 Administrative Log
2 Auto Daylight Saving
3 Customizable Time Zone
4 Front Panel LCD Setup
5 FTP Server
6 HTTP server
7 HTTP SSL support
8 Multiple configuration
9 NTP time synchronization
10 Password Security
11 RS232
12 System Event Log
13 Telnet
14 Time Zone Support
15 User Account Manager
16 Web-based GUI
17 Web-based Real Time Monitor
18 Web-based Voice File Management
Network Management
1 DHCP
2 Fixed IP
3 DNS
4 Dynamic DNS
5 Ping
6 TOS field setting
7 SNMP V2 MIB I & II
8 SNMP get command
9 SNMP set command
10 SNMP Trap
11 H.341 MIB Support
12 SysLog Support
H.323 Protocol Support
1 H.323 V3
2 H.323 ID
3 E.164 ID
4 Fast Connect
5 H.450
6 H.245 Tunneling
7 Early H.245
8 Cause Code Mapping
1 Cause Code Mapping
2 HTTP Digest Authentication
3 SIP Call on Hold
4 SIP Early Media
5 SIP Overload Redirect
6 SIP Transfer (unattend)
7 SIP Transfer (attend)
8 SIP/TCP
9 SIP/UDP
7 H.323 to SIP FAX Relay
7 PSTN to H.323 Call
8 PSTN to PSTN Call
9 PSTN to SIP Call
10 SIP to H.323 Call
11 SIP to PSTN Call
12 SIP to SIP Call
13 SIP to SIP Fax Relay
14 VoIP to VoIP RTP unRouted
15 VoIP to VoIP RTP Routed
Enhance Service
1 ANI Access List
2 DNIS Access List
3 DID/DOD
4 PSTN Two Stage Dialing
5 VoIP Two Stage Dialing
6 Intelligent PSTN Call Routing
7 In-trunk hunting method
8 Ring Back Tone Generation
9 Call Progress Tone Support
10 Web-based Call Flow GUI
11 Play Credit Time Duration
12 Play Credit Balance
13 Almost-time-expired notify tone
14 IVR for PSTN
15 IVR for SIP
16 IVR for H.323
17 IP Access List
18 ANI Replacement
19 DNSI Replacement
AAA
1 Call detail record (CDR)
2 RADIUS Authentication
2 RADIUS Authorization
3 RADIUS Accounting
4 Redundant RADIUS Server Support
5 PSTN Prepaid Support
Drag and Drop interface, Full
control of call behavior (one-stage
or twoSupport time duration play back
(Chinese & English), Power call
information branch, Collected
information validation, Active
disconnect & reconnect without
hang up, Selected disconnect
cause code & behavior
1 Embedded Prepaid Service
2 Embedded Postpaid Service
3 Point/second Calculation
4 Second/point calculation
5 Auto Disable/Clean User
6 PSTN Prepaid Support
7 VoIP Prepaid Support
System Limitation
1 Max DM
2 Max IP ACL
3 Max DNIS ACL
4 Max ANI ACL
5 Max User ACL
6 Max Phone Book Entries
7 Max Call Flow Component
8 Max CDR Keep Days
9 Max Voice File Storage
Manual
1 English User Guide
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
4096
2048
4096
4096
20000
10000
256
5
10 hours
Yes
VIP-2100 User’s manual - 9 -
1
2
3 4 5 7 6
10
1
2 3 7 4 5 8 6
VIP-2100 Appearance Description
VIP-2100 Front Panel:
Functions:
1: Power LED
2: Network1 Interface LED
3: Network2 Interface LED (not used)
4: H/D LCD
5: Power Switch
6: System Status LED
7: LCD Panel
8: LCD Touch Panel
VIP-2100 Rear Panel:
Functions:
1: Electric Fan
2: AC Power outlet
3: AC Power switch (Keep on)
4: Trunk E1/T1 port
5: VoIP Ethernet port
6: Keyboard/Mouse
7: Com1 port
8: Ethernet port
9: VGA
10: print port (not available)
VIP-2100 User’s manual - 10 -
8
9
Chapter 2 Logon VIP-2100
After connected E1/T1 & Ethernet cables into the VIP-2100, 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 setup VIP-2100, you need to have
Java Standard Runtime (1_4_1_02) to make it work. Please refer to
Appendix 2 Java plug-in Install for detail.
Logon VIP-2100
Setp1: Start IE5.0 (or later version) to navigate VIP-2100 Management
System by typing the default IP address (the default URL is
http://192.168.111.111:10087). The screen will display User ID and
Password as figure 2.1-1.
Figure 2.1-1
☻Note: The default network IP address is 192.168.111.111 and subnet
mask is
255.255.0.0
Step 2: Enter log 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
VIP-2100 User’s manual - 11 -
Step 3: The screen shows the Home Page of VIP-2100 as figure 2.1-3.
Figure 2.1-3
Network Configuration
Step 1: After successfully logon to the system, we need to change the
network configuration. Click Control→Network to setup the network
parameters as figure 2.2-1.
Figure 2.2-1
VIP-2100 User’s manual - 12 -
Step 2: Enter the deserved IP address, Submask and default gateway. Apply
the change by clicking apply button as figure 2.2-2.
Figure 2.2-2
Step 3: When screen shows “Setup network configuration successfully!”
It means the IP Network setting is successfully changed as figure 2.2-
3.
Figure 2.2-3
☻Note: “Network Control” takes around 5-second to apply the new
network configuration. Please logon again with new IP address
after 5 seconds.
System Time Configuration
Step 1: When re-logon to the new IP address; the next is to setup the system
time zone. Click Control→System Time Zone to setup the system
as figure 2.3-1.
Figure 2.3-1
Step 2: After apply the new time zone, click Back to adjust the date and time
as figure 2.3-2.
VIP-2100 User’s manual - 13 -
Figure 2.3-2
Step 3: Enter current date and time. Apply the change by clicking Apply
button as figure 2.3-3.
Figure 2.3-3
Step 4: The screen will shows “Setup system time successfully!”It means
the System Time setting is successfully changed as figure 2.3-4.
Figure 2.3-4
Step 5: If you would like to use SNTP to sync time with a SNTP V4 Server,
click Time Sync button to setup it as figure 2.3-5.
Figure 2.3-5
Account Manager
Step 1: You can manage your user account by click Control→Account
Manager. Add a new user account, Click New button as figure 2.4-1.
Figure 2.4-1
Step 2: Enter the new user ID, password, user role and description, as you
need. Apply the change as figure 2.4-2.
VIP-2100 User’s manual - 14 -
Figure 2.4-2
Field Description:
• User ID: Login User ID
• Password: Login Password
• Confirm Password: Confirm new password again
Step 3: When screen shows “Create user account successfully!” It means
user account setting is successfully created as figure 2.4-3
Figure 2.4-3
☻Note: The system provides 2 USER ID by default:
User 1: “root” Password: “root”
User 2: “admin” Password: “admin”
VIP-2100 User’s manual - 15 -
Relogin
Step 1: Click Control→Relogin to relogon by another user account as figure
2.5-1.
Figure 2.5-1
Step 2: Enter new User ID and Password to relogon the VIP-2100 as figure
2.5-2.
Figure 2.5-2
Step 3: The screen shows the Home Page of VIP-2100 as figure 2-5-3.
This section is going to setup the VoIP interface.
Step 1: Now we are going to setup the VoIP interface, click Configuration→
Interface to setup VoIP T1/E1 interface as figure 3.1-1.
Figure 3.1-1
Step 2: Double-click the installed interface (i.e Interface ID:0) to config it as
figure 3.1-2.
Figure 3.1-2
Step 3: Modify the VoIP Interface parameters (i.e. IP Address, Protocol Tag,
Subnet Mask and Default gateway) and apply the change by clicking
Apply as figure 3.1-3.
Figure 3.1-3
VIP-2100 User’s manual - 18 -
Frequency changed parameters: (Refer to section “Interface
Configuration”for more detail)
• IP Address: 192.168.19.174
• Subnet Mask: 255.255.255.0
• Default Gateway: 192.168.19.254
• PCM Type: A-law or Mulaw
☻Caution: Subnet Mask does not support Supernet.
Step 4: After successfully to change the Interface configuration, the screen
come back the page of Interface Configuration as figure 3.1-4.
Figure 3.1-4
T1/E1 Trunk Configuration
This section is going to setup the PSTN trunk parameters.
Step 1: Select the installed interface to modify the trunk parameter by click
Detail button as figure 3.2-1.
Figure 3.2-1
VIP-2100 User’s manual - 19 -
Step 2: Select the trunk to be modified, and click Modify button as figure 3.2-
2.
Figure 3.2-2
Step 3: Modify the trunk parameters (i.e. Trunk Type, Termin Side, Trunk
Mode, Protocol Tag, Line Code) and apply the change by clicking
Apply as figure 3.2-3.
Figure 3.2-3
Frequency Changed Parameters:
• Trunk Type: E1 or T1
• Termin Type: User Side or Network Side
• Trunk Mode: Normal
• Protocol Tag: ISDN protocol used
• Line Code: T1 or E1 line code used
VIP-2100 User’s manual - 20 -
Step 4: After modifications are made to the Trunk Configuration, the screen
comes back the page of Trunk Configuration as figure 3.2-4.
Figure 3.2-4
H.323 Configuration
This section is going setup the H.323 parameter. If you only need SIP
calls, you can skip it.
Step 1: Click Configuration→H.323 to setup the H.323 parameters for
Gatekeeper related information as figure 3.3-1.
Figure 3.3-1
Frequency used parameters:
• Register to Gatekeeper: Yes
• Gatekeeper IP: 192.168.5.1
• E.164 Tel: 113
• Register H.323 ID: 113
VIP-2100 User’s manual - 21 -
Step 3: You can see the screen display the new configuration of the H.323
Configuration as figure 3.3-3.
Figure 3.3-3
SIP Configuration
This section is going setup the SIP parameter. If you only need H.323
calls, you can skip it.
Step 1: Click Configuration→SIP to setup the SIP parameters for SIP Proxy
Server related information as figure.3.4-1.
Figure 3.4-1
Frequency used parameters:
• SIP Register: Yes
• Primary Registar Server: 192.168.19.150
• Primary Registar Port: 5060
• Primary Registar User: 173
• Primary Registar Password: 173
VIP-2100 User’s manual - 22 -
• Primary Outbound Proxy Server: 192.168.19.150
• Primary Outbound Proxy Port: 5060
• Primary Outbound Proxy User: 173
• Primary Outbound Password: 173
Step 3: You can see the screen display the new configuration of the SIP
Configuration as figure 3.4-2.
Figure 3.4-2
Digit Manipulation
The purpose of “Digit Manipulation” is to add or drop dialed digits for
PSTN or IP side (Interface configuration for PSTN side & H.323 Configuration
for IP side) at the selected interface in order to meet local PSTN dialing
requirement. It can also be used in Call Flow Edit for flexible usage.
Step 1: We introduced the group and interface dependent digital manipulation
to meet the customer’s requires. Click Digit Manipulation to add a
new Digit Manipulation Group, add as figure 3.5-1.
Figure 3.5-1
VIP-2100 User’s manual - 23 -
Step 2: Enter the related parameters and click Apply button as figure 3.5-2.
Figure 3.5-2
Field Description:
• Group ID: 0 (DM Group identify)
• Description: H.323: H323 In Drop
SIP: SIP In Drop
Step 3: Click the New created DM and Detail button to add digits setting as
figure 3.5-3.
Figure 3.5-3
Step 4: Click New button to add a new DM rule as figure 3.5-4.
Figure 3.5-4
VIP-2100 User’s manual - 24 -
Step 5: Create a new H.323 DM Group “1” and DM detail is show as follows:
Figure 3.5-5
H.323 Incoming Call DM Setting:
• Matched Pattern: 5 (pattern to be matched)
• Group ID: 1-H323 In Drop (belong to this DM group)
• Drop: 5 (drop digits)
H.323 incoming call
↓
Dialed number: 582265699
↓
Match the pattern 5
↓
Delete 5 (Drop)
↓
New dialed number becomes 82265699
Step 5: Also create a new SIP DM Group ‘2” and DM detail is show as follows:
Figure 3.5-6
SIP Incoming Call DM Setting:
• Matched Pattern: 113 (pattern to be matched)
• Group ID: 1-SIP In Drop (belong to this DM group)
VIP-2100 User’s manual - 25 -
• Drop: 113 (drop digits)
SIP incoming call
↓
Dialed number: 11307688222
↓
Match the pattern 11307
↓
Delete 113 (Drop)
↓
New dialed number becomes 07688222
Step 6: Create a PSTN incoming call DM Group “3” and DM detail is show as
follows:
Figure 3.5-7
PSTN DM Setting:
• Matched Pattern: 0282265699 (pattern to be matched)
• Group ID: PSTN In Drop (belong to this group id)
• Drop: 0282265699 (drop digits)
PSTN incoming call (DNIS mode)
↓
Dialed number: 02822656991001
↓
Match the pattern 0282265699
↓
Delete 0282265699 (Drop)
↓
New dialed number becomes 1001
☻Note: Digit Manipulation have to tapped for PSTN Side (Trunk
Outbound/Inbound DM Group), VoIP Side (VoIP
→
→
VIP-2100 User’s manual - 26 -
Outbound/Inbound DM Group) or Call Flow (refer to section “Call
Flow Editor”) to take effect.
Chapter 4 Call Flow Editor
Call Flow Editor is used to control the call behavior including voice
prompt, AAA, DM…etc. It requires Java run time to run.
Step 1: Click Control→Call Flow Editor to create a Call Flow, click
button to activate IVR Tool as figure 4-1
Figure 4-1
Component Description:
• New: Create a new call flow
• Load Call Flow: Load call flow from VIP-2100
• Save: Save a call flow in VIP-2100
• Cut: Cut a component
• Copy: Copy a component
• Paste: Paste a component
• Delete: Delete a component
• Line: Connecting 2 components together
• Select: Select the component at call flow workspace
• Scroll: Scroll the call flow workspace
• Zoom: Zoom in or zoom out the workspace
• View Grid: View or not
• Show Component Table: Show all component table
VIP-2100 User’s manual - 27 -
Step 2: Drag and prop the required component icon into the workspace as
figure 4-2.
Figure 4-2
Right click the component to bring up the component propriety to
setup parameter:
• AAA: Send Authorization or Authentication for validation
o Type: AAA type selection
- Authorization: Send RADIUS Authorization packet out
- Authentication: Send RADIUS Authentication packet out
Success To: Success to component
Failed other to: Failed to component
VIP-2100 User’s manual - 28 -
o Failed Reason: Return code from RADIUS server
o Line Propriety:
- Invalid Account
- Account In Use
- Zero Balance
- Account Expired
- Over Credit Limit
- Number of Retries Exceeded
- Insufficient Balance
J Note: Detail response attributes, please refer RADIUS Format
Attributes
• Answer: Answer incoming call (PSTN only)
• Branch: Play an announcement and branch into different route
o Voice File: Voice prompt file (“. raw” format) to be playing
o DTMF Length: Number of DTMF to be receiving
o Others: Default flow if not match
o DTMF: DTMF match pattern
o Goto: The next component if matched
o Line Propriety:
-Branch Line: DTMF branch line setting
• CDV: Collected Digit Validation
VIP-2100 User’s manual - 29 -
o Check Parameter: Check parameter type (DNIS, ANI….)
o Digit From: Start digit from
o Digit To: End digit to
o Valid To: If the checked variable is success to validate
o Invaried To: If the checked variable is not success to validate
• CIB: Call Information Branch
o Info Type: Information type selection
- ANI: Calling Number
- DNIS: Called Number
- IP: IP Address or network (e.g. 192.168.0.0)
- PSTN: E1/T1 trunk and channel filter, format: interface id-
trunk id- trunk start- trunk stop
- Prefix: The prefix to be match
0-1-17-31:
0: Interface ID (Always 0)
1: Trunk ID: 1
17: Start from B Channel 17
31: Stop from B Channel 31
o Goto: The component to run next
o Call Info Branch Line: ANI, DNIS, IP or PSTN goto setting
• CIV: Call Information Validation, the user need setup the ACL for
DNIS and IP TO take effect
o Info Type: The infor type to be validation
VIP-2100 User’s manual - 30 -
-DNIS: Called number
-ANI: Calling number
-IP: In coming IP address
-User: User ID
o Allow To: If it is met the ACL defined
o Disallow To: If it is not met the ACL defined
• CTB: Call Type Branch
o PSTN To: Route for PSTN call
o H.323 To: Route for H.323 call
o SIP To: Route for SIP call
• Cut Rule: Cut a system variable into different parts
o Cut From: Cut start digit from (start from 1)
o Cut To: Cut end digit to
o Assign To: Store the cutted result into
• Disconnect: Disconnect the call
• DM: Digit Manipulation
VIP-2100 User’s manual - 31 -
o DM Parameter: Manipulation ANI or DNIS
o DM Group ID: Apply to DM group
• MakeCall: Make Call to PSTN or H.323/SIP site
o Route Mode: Gatekeeper Call or P2P Call or PSTN…etc. (for PSTN
incoming call, please select the Gatekeeper, P2P Call, or SIP Proxy
callTA;for H.323/SIP incoming call, please select the PSTN call)
o Transport Address: When used for “H.323 TA” routing mode, the
format used is “Ipaddr:port” (e.g. 192.168.111.50:1720)
o Active Disconnect: Enable PSTN user can actively disconnect the
call or not
o Active Disconnect Digit: The DTMF digit to be tread as the
disconnect trigger. Only can be used “Active Disconnect” enable
o Active Disconnect To: The next component when active disconnect
is occurred
o Inter Digit Timeout: The max time to in seconds to wait between two
digits.
o RTP Route: Voice RTP routing over VIP-2100 or not, for VoIP to
VoIP call
o Finish To: Successfully connect to remote site
o Failed Other to: The next component when default failed call
o Failed Reason: Failed reason selection
o Failed To: When the failed reason occurred go to
VIP-2100 User’s manual - 32 -
o Line Propriety:
-PSTN: PSTN disconnect reason code:
-Normal Call Clear
-User Busy
-No User Response
-No Answer
-Call Reject
- VoIP: VoIP disconnect reason code:
-User Busy
-No Answer
-Unreachable
-Other
• PA: Play Announcement
o Dynamic Play: Dynamic play voice file by combine prefix and
variable as the file name
o Enable: Combine prefix to variable as the voice file to play
-Prefix: Voice file prefix (e.g. prefix: WT, variable: user1 (contact 201, played voice file is “WT201.raw”)
-Variable: Variable to be appending as the voice file name
o Disable: Use filter voice prompt file
-Voice File: Voice prompt file
o Interrupted: Voice can be interrupted or not
• PB: PlayBalance for prepaid purpose
o Voice File: Voice prompt file
o Language: Play balance language section
-English
-Chinese.
o Interrupted: Voice can be interrupted or not
VIP-2100 User’s manual - 33 -
• PCUI: Prompt and Connect User Information
o Play Type: Dial tone or voice prompt selection
o Voice File: Voice prompt file
o Max DTMF: Maxtor of DTMF to be received.
o Assign To: Result (received DTMF) will be assign to
o End of DTMF: The digit to indicate dial end.
o Interrupted: Voice can be interrupted or not
• PD: Play Duration for prepaid purpose
o Voice File: Leading voice prompt file
o Language: Play duration language section
-English
-Chinese
o Interrupted: Voice can be interrupted or not
☻Note: The RADIUS servers need to be setup to send H.323/SIP
credit time or internal RADIUS must be used.
• PSTN L.H: PSTN Line Hunting
o Success To: If fine an available channel by system setup (call
hunting)
o Failed To: If not fine an available channel by system setup (call
hunting)
VIP-2100 User’s manual - 34 -
• Set Data: Assign value to a variable
o Assign To: Assigned variable
o Use SysParam: Use system parameter to replace or not
o Value: ANI, DNIS, User ID or other digits
Configuration Management provides a way to save and reload the
system configuration for future use.
VIP-2100 User’s manual - 39 -
Load a Configuration:
Step 1: When you need to load a saved configuration, click a saved
configuration (i.e. 04/26/2004 Loading Test) item to load it back as figure
4.1-1.
Figure 3.7-1
Step 2: When screen shows “Current configuration will lost! Are you sure
to load this configuration?”click on OK button to load he saved
configuration to the working configuration as figure 4.1-1.
Figure 3.8-2
JNote: It is need to restart the system to take effect of the new-loaded
working configuration.
Save the working Configuration:
Step 3: To save the current configuration, select a new created configuration
and click Save button, when screen shows “Description”, please
enter the configuration description (i.e. Billing Test) for the saved
configuration as figure 4.2-2.
Figure 3.8-3
Step 4: You can see the screen display the changes as figure 4.2-4.
VIP-2100 User’s manual - 40 -
Figure 3.8-4
Backup the working configurations:
Step 5: To backup the running configuration, click on Backup button, to back
up local hard disk. The whole running configuration will be compress
into a zip file (file name: export.zip) and transfer back to local as figure
4.2-2.
Figure 3.8-5
Restore configuration:
Step 6: To restore the backup configuration file, click on Restore button,
when screen shows “Import Configuration file”, select backup file
(i.e. c:\export.zip) click on Import button to restore the configuration to
the working configuration as figure 4.2-2.
Figure 3.8-6
Compact the database file:
Step 7: In order to optimize the system performance, you can optional
compact the database by click Compact button as figure 4.1-2.
Figure 3.8-7
J Note: Please make sure that there is no others person to use database
concurrently.
VIP-2100 User’s manual - 41 -
Apply Change
When you load a new working configuration, the system must be
restarted to take effect.
Step 1: Click Configuration→Apply Change, the screen show “ The
change you mode need to restart the system for apply please
confirm to restart or do it later.”Click on OK/Cancel to restart the
• CDR Mode: Call detail record generating mode (Refer to “Appendix 3
Retrieve CDR Information” for detail file description)
o File Only: Log CDR into the file only. It can be retrieved by ftp
(directory c:\cd cdr).
o Radius Start/Stop: Log CDR into the file and send RADIUS start/stop
billing message out.
- VoIP: enable VoIP site RADIUS billing message or not.
- PSTN: enable PSTN site RADIUS billing message or not.
o Radius Stop: Log CDR into the file and send RADIUS stop billing
message out.
- VoIP: enable VoIP site RADIUS billing message or not.
- PSTN: enable PSTN site RADIUS billing message or not.
• CDR Keepdays: CDR system keeping days
• Hot Swappable: Hot swappable support (reserved)
• First Digit Timeout: The max to time (in second) waits for receiving the
first digit entered (5~20 sec).
• Inter Digit Timeout: The max to time (in second) waits for the between
two digits (5~20 sec).
• Debug Level:
o Critical: Show critical error messages only
o Warring: Show warring and critical error message only
o Information: Show information, warring and critical message only
o Debug: Show all debug messages
o Full Trace: Show all status and debug messages
☻Note: Please set to “Critical” only, or the whole system performance
will be hitted.
VIP-2100 User’s manual - 52 -
• Time Expired Notify: Seconds to be notifying caller before
communication expired. This function is used for Pre-Paid calling card
service and must cooperate with RADIUS Server.
• Almost Expired Tone: Communication expired notice tone selection
• Fast Response Timeout: The maximum times to wait for response. It’s
depended on the network speed.
• No Answer Timeout: The maximum the (in second) to wait the remote
party answer (pick up phone)
o Notify Tone#1:
o Notify Tone#2:
• Authentication Mode: Authentication by VIP-2100 or RADIUS
o Internal: Authentication building User ACL
o External: Authentication by RADIUS
o Ext. AAA Failure Opt: Bypass or disconnect incoming calls when
external
• Version: 5.1
Interface Configuration
Start Path: Configuration→Interface
Figure 7.2-1
Basic Parameter Description:
• Interface ID: System parameter
• Card slot: System parameter
• Interface Type: System parameter
• Description: System parameter
• Serial No: System parameter
• License Key: System parameter
• IP Address: IP address used for voice RTP stream
• Subnet Mask: Submask (doesn’t support super class)
• UDP Port Base: UDP port used for RTP stream, each channel needs 3
RTP ports and must be started by a multiple of 10
• IP Precedence: Voice package priority setting
o Routine Precedence
o Priority Precedence
o Immediate Precedence
o Flash Precedence
o Flash Override Precedence
o Critical Precedence
o Internetwork Precedence
o Network Precedence
• IP TOS: Top of Service with the following priority selection
o Normal Service
o Minimize Monetary
o Maximize Reliability
o Maximize Thought
o Minimize Delay
• PCM Idle Pattern: This pattern will be sending on each B channel PCM
time slot when the channel is idle (not connected). The default value
for A-Law is 0xff and for Mu-Law is 0x55. You only change it when
SWITCH need.
• CAS Idle Pattern: When channel is idle, ABCD (CAS) pattern to be
applied CAS signaling bus
• Jitter Min Delay: The minimum delay time of Jitter buffer. The range is 0
to 150ms. Default value is 150ms. Which has better voice quality but
the delay time will be long.
• Jitter Opt Factor: Jitter buffer optimization factor from 0 to 12. The
default value is 7. Set to 0 will have lowest voice delay but have bad
VIP-2100 User’s manual - 54 -
voice quality. Set to 12 will have long voice delay but with better voice
quality
• EC Tail Length: Echo Cancellation Length, default value is 25ms
• Silence Compress: Enable silence compress or not
• TDM Bus Clock: TDM Bus clock source
o Internal: derived from internal oscillator
o External: derived from external PSTN E1/T1 clock
Dial Plan Configuration
Dial Plan can be used to assign the ISDN number plan based on prefix
setting.
Start Path: Configuration→Interface→Dial Plan
Figure 7.3-1
Basic Parameter Description:
• Prefix: Called party number prefix
• Src Num Plan: ISDN Source number plan
• Src Num Type: ISDN Source number type
• Dest Num Plan: ISDN destination number plan
• Dest Num Type: ISDN destination number type
• ApplyTo: Trunks apply to
T1/E1 Trunk Configuration
Start Path: Configuration→Interface→Trunk
VIP-2100 User’s manual - 55 -
Figure 7.3-1
Basic Parameter Description:
• Interface ID: System parameter
• Trunk ID: System parameter
• Trunk Type: T1or E1 selection
• Description: Description for this trunk ID
• Termin Side: Network site or User Site (normally, you set to “user site”
when connect to switch)
o User Side
o Network Side
• Trunk Mode: Trunk operation mode
o Disable: Disable the trunk
o Normal: Accept PSTN and VoIP calls
o PSTN incoming only: Allow the PSTN incoming calls only
o H.323 incoming only: Allow the H.323 incoming calls only
• Hunting Method: PSTN trunk hunting method for available channel
o Random: Hunt randomly
o Cyclic: Initial hunt (after power-up/reboot) begins with B channel 1;
subsequent hunts begin with position following last successfully
allocated resource
o Rotary: Hunt always begins with B channel 1
o Reverser Rotary: Hunt always begins with B channel 31
o Reverser Cyclic: Initial hunt (after power-up/reboot) begins with B
channel 31, follows next available channel in reverser order
• CAS Variance: CAS counting variance
• Framing Method:
o For T1
- super frame
- 4-frame multi-frame
- 12 frame multi-frame (D4)
- extend super frame without CRC6
- extend super frame with CRC6
- 72-Frame Multi-Frame
o For E1:
- Automatic CRC4 or Double Frame selection
- Double Frame Format
- CRC4 multi-frame
- CRC4 extend multi-frame
• Protocol Tag: supported protocol on T1/E1 interface with PSTN switch
o For T1:
- T1 CAS
- T1 RAW CAS
- T1 NI2 ISDN
- T1 4ESS ISDN
- T1 5ESS 9 ISDN
- T1 5ESS 10 ISDN
VIP-2100 User’s manual - 56 -
- T1 DMS100 ISDN
- T1 NTT ISDN: used to connect NTT INS-1500 ISDN standard (Japan
Only)
- T1 HKT ISDN
- T1 QSIG
- T1 EURO ISDN
- T1 DMS100 MERIDIAL ISDN
- T1 NI1 ISDN
o For E1:
- E1 EURO ISDN: used for most of European ISDN standard
- E1 MFCR2
- E1 CAS
- E1 RAW CAS
- E1 AUSTEL ISDN: Australia E1 ISDN Variance
- E1 HKT ISDN: Hong E1 ISDN Kong Variance
- E1 KOR ISDN: Korea E1 ISDN Variance
- QSIO
- E1 TNZ ISDN
• Line Code: T1: you can choose AMI, B8ZS; E1: you can choose AMI,
HDB3
• PSTN Trace: PSTN layer debug trace. It will generate a debug trace file
for tracing purpose. Only enables it under Welltech technical supports
instruction and disable it when complete the debug
• Inbound DM Group: Digit Manipulation group used for incoming calls
associated to this trunk
• Outbound DM Group: Digit Manipulation group used for outgoing calls
• Local Ring Back: Provide ring back tone for PSTN or not. It only works
when VoIP outgoing Fast Start is disabled.
• Channel Mask: Channel mask for incoming or outgoing calls (default:
0xffffffff)
Start from MSB each bit, indicate a time, slot a trunk (e.g. 0x0000ffff:
0~15 B channel mask, 17~31 B channel free)
• Input Gain: Voice Gain from IP to PSTN side (default: 0 db)
• Output Gain: Voice Gain from PSTN to IP side (default: 0 db)
• Q.931 General Opt.: used for Q.931 general behavior.
o 0x0001: No Status message send for unknown facility IE if it is set
o 0x0002: No Status message send for invalid content of a valid
facility IE if it is set
o 0x0080: Send Connect Ack message when receive Connect
message if it is set, you can OR the required option together
• Q.931 Incoming Opt.: used for Q.931 incoming call behavior
o 0x0800: include Channel-ID IE in the first reply message (e.g. Call
Proceeding or Alerting)
o 0x2000: enable the system to include Channel-ID IE in the Call
Proceeding message, you can OR the required option together
• Q.931 Outgoing Opt.: used for Q.931 outgoing behavior
o 0x0010: use Mu-law if this bit is set, or A-law will be used. Apply
only for Korea variance, you can OR the required option together
• Trans Cap: Transfer Capability
o Voice Service
o Data Service
o Modem Service
• CallID Transfer Type: Call ID transfer type
o Disable Caller ID: default parameter
o Transparent Caller ID
o Relay Caller ID
o Bypass Caller ID
VIP-2100 User’s manual - 58 -
Rest Configuration
Reset a channel or a trunk idle state.
Start Path: Configuration→Interface→Detail→Reset
Figure 7.4-1
Start Path: Configuration→Interface→Detail→Reset
Figure 7.4-2
Basic Parameter Description:
• Trunk: Reset trunk ID
• Channel: Rest channel selection
- All Channel: Reset all channel
- 0~31: Reset 0~30 logical channel to reset
H.323 Configuration
Start Path: Configuration→H.323
VIP-2100 User’s manual - 59 -
Figure 7.5-1
Basic Parameter Description:
• Register To Gatekeeper: Register to Gatekeeper or not
o Yes: Register to GK
o No: Not register to GK
• Gatekeeper IP: Gatekeeper IP Address
• Gatekeeper RAS: UDP Port number listened on Gatekeeper (default:
1719)
• E.164 Tel: Telephone number to be registered to Gatekeeper
• Register H.323 ID: H.323 alias name to be registered to Gatekeeper
• Register Time To Live (sec): The registration maximum time to live
setting when registered to the Gatekeeper
• Response Timeout (Q.931)(sec): The maximum time to wait for
response from sending call setup signal out
• Connect Timeout (Q.931)(sec): The maximum time to wait for
connection (answer) from dialing out the destination number
• DTMF Relay: DTMF transfer type selection
o RTP relay (RFC 2833): DTMF relay via RTP packet (RFC2833
standard)
o DTMF transparent: transmitter DTMF over voice channel
o H.245 Signal input: DTMF relay via H.245 user signal input
o H.245 Alphanumeric: DTMF relay via H.245 Alphanumeric signal
o Q.931 User Information: DTMF relay via Q.931 User to user
• Max TCP Connection: Max Call: The maximum SIP TCP calls.
• Outbound Use TCP: Use SIP TCP for outbound call or not. If it set to
no, UDP is used.
• Register Use TCP: Use SIP/TCP to register to SIP register.
• TCP Port: The local TCP port on which the SIP Stack listens.
• UDP Port: The local UDP port on which the SIP Stack listens.
• Reliable Provision: Support PRACK or not (100rel)
• Max Call Leg: The maximum number of call-legs the SIP Stack
allocates. You should set this value to the maximum number of call
your expect the SIP Stack to handle simultaneously.
• Max Transaction: The maximum number of transactions the SIP Stack
allocates. You should set this value to the maximum number of call
your expect the SIP Stack to handle simultaneously.
• Max Register Client: The maximum number of Register-Clients the SIP
Stack allocates. You should set this value to the maximum number of
call your expect the SIP Stack to handle simultaneously.
• Send Receive Buffer Size: The buffer size used by SIP Stack for
receiving and sending SIP messages.
• Reject Unsupported Extension: Yes or No
• Message Pool Page Size: Used to hold and process all incoming and
outgoing message in the from of encoded messages or message
objects. It is recommended that you configure the page size to the
average message size your system is expected to message.
• General Pool Page Size: Used by SIP Stack objects, such as call-legs
and transaction, to store the internal fields. For example, the call-legs
object will store the To, From and Call-ID headers and the local and the
remote contact addresses on the general pool pages. The general pool
is also used from other activities that demand memory allocation.
VIP-2100 User’s manual - 64 -
• Application Pool Page Size: The size of page in the application pool
• Retransmission T1: T1 determines several timer 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: 500)
• Retransmission T2: 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: 4000)
• Retransmission T4: 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: 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: 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.
(Default Value: 32000)
• Provisional Timer: When a client receives a provisional response, it
continues to retransmit the request, but with an interval of
provisionalTimer milliseconds.
• Cancel General No Response Timer: When sending a CANCEL
request on a General transaction, the User Agent waits
cancelGeneralNoResponseTimer milliseconds before timeout
termination if there is no response for the cancelled transaction.
• Cancel Invite No Response Timer: 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.
• General Request Timeout Timer: 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)
• 183 to Alerting: When receive a SIP 183 message from remote site,
send Alerting in stead of Call Progress Indicator
• AutoSend 183: VIP-2100 always send Call Progress Indicator (SIP 183)
to VoIP party. It can be used for CAS protocol to enable early media.
• Behind NAT: Does VIP-2100 is located behind NAT or not
• Public Signal IP: The static mapped IP for SIP signal
VIP-2100 User’s manual - 65 -
• Public Signal Port: The static mapped Port for RTP stream
• Public RTP IP: The static mapped RTP IP
• Public RTP Port: The static mapped RTP starting port
• Public RTP Port Interval: The VIP-2100 has at least 30 RTP channels.
Each channel needs 3 ports mapping on NAT Server. The interval is
used to caculate the right port for each channel.
• Overload Redirect: SIP overload redirect when VIP-2100 is not able for
service the call
• Redirect Host: Redirect host URI (format: user@siphost, siphost)
• Redirect Port: Redirect port number
• Send 487 When Recv CANCEL: When receive CANCEL form remote
site, send “487 Request canceled” or not
• Caller ID Mode:
o Local: use VIP-2100 proxy user id
o Caller: use SIP calling party ANI
• Receive Hold music source:
o Auto: Auto determinate to play hold tone based on SIP signaling.
o Local: Play hold tone locally.
• On Hold music: Hold tone music file name
VIP-2100 User’s manual - 66 -
Behind NAT Example 1:
VIP-2100 NAT Server Setting
One-by-One Static
IP Mapping
192.168.111.112 210.59.163.11
Static Port Mapping 192.168.111.111:5060 210.59.163.10:10000
VIP-2100 NAT Enable Setting:
Public Signal IP: 210.59.163.10
Public Signal Port: 10000
Public RTP IP: 210.59.163.11
Public RTP base port: 4000 (same as “Interface→Advance’s Config”)
Public RTP Port Interval: 10
Behind NAT Example 2:
VIP-2100NAT Server Setting
Static Port Mapping 192.168.111.111:5060 210.59.163.10:5060
RTP Channel 01 192.168.111.112:4000
4001
4002
RTP Channel 02 192.168.111.112:4010
4011
4012
.
.
.
.
.
.
RTP Channel 30 192.168.111.112:4310
4311
4312
210.59.163.10:10000
10001
10002
210.59.163.10:10003
10004
10005
.
.
.
210.59.163.10:10357
10358
10359
VIP-2100 NAT Enable Setting:
Public Signal IP: 210.59.163.10
Public Signal Port: 5060
Public RTP IP: 210.59.163.10
Public RTP base port: 10000 (same as “Interface→Advance’s Config”)
Public RTP Port Interval: 0
VIP-2100 User’s manual - 67 -
Access Control
Access Control list can be used to filter the calls forms the IP Network,
DNIS, and ANI. It must be used in call flow edit to take effect.
IP ACL
Start Path: Configuration→Access Control→IP ACL
Figure 7.7-1
Parameters:
• IP Network: IP Address or prefix used to be filtered
• Access Mode:
o Allow: the inputs IP Network are allowed for calls.
o Disallow: The inputs IP Network are disallowed for calls.
☻Note: If in the system has both allowance and disallowance setup, the
system will check allowance first and disallowance later. If only
disallowance inputted all IP will allow to work except disallowed
network. If only allowance inputted, only those IP from
allowance list will work.
ANI ACL
ANI ACL
Start Path: Configuration→Access Control→ANI ACL
Figure 7.7-2
Parameters:
• ANI: Calling party number used to filter
• Access Mode:
o Allow: the calling numbers are allowed for calls
o Disallow: The calling numbers are disallowed for calls
☻Note: If in the system has both allowance and disallowance setup, the
system will check allowance first and disallowance later. If only
disallowance inputted all ANI will allow to work except
disallowed ANI. If only allowance inputted, only those ANI from
allowance list will work.
VIP-2100 User’s manual - 68 -
DNIS ACL
Start Path: Configuration→Access Control→DNIS ACL
Figure 7.7-3
Parameters:
• DNIS: Called party number used for filter
• Access Mode:
o Allow: The called numbers are allowed for calls
o Disallow: The called numbers are disallowed for calls
☻Note: If in the system has both allowance and disallowance setup, the
system will check allowance first and disallowance later. If only
disallowance inputted all DNIS will allow to work except
disallowed DNIS. If only allowance inputted, only those DNIS
from allowance list will work.
User ACLUser ACL is used to store subscriber information when internal AAA is
• Prepaid Point: Allowed prepaid point (When prepaid point is used, the
system will deduct it automatically base on the rate setting.)
o Postpaid: postpaid user
• Status:
o Active: User is activeled
o Inactive: User is inactived
☻Note: 1. IP Authentication method must be set to “ internal AAA” to
talk effect.
VIP-2100 User’s manual - 69 -
New a Calling Rate: The calling rate will have different appearance for
different calling rate policy set in Radiu configuration.
Click Calling Rate button to add a new calling rate as figure 7.7-5.
Figure 7.7-5
Point per Second calling rate:
Calling rate (point per second) is used to convert prepaid point into
prepaid time in second. For example, you can set calling rate to 5 for “100”
prefix. When a caller, which has 200 prepaid point, calls “100xxxx”, the max
talk time will be 200/5=40 seconds. If a calling rate is set to “0”, it means free
charge.
New a Calling Rate (Second per Point):
Click Calling Rate button to add a new calling rate as figure 7.7-6.
Figure 7.7-6
Second per Point calling rate:
. Calling rate (Second per point) is used to convert prepaid point into
prepaid second in time. For example, you can set calling rate (Second) to 6,
charge point to 1 for “113” prefix. It means that every 6 seconds charge 1
point. When a caller, which has 200 prepaid point, calls “113xxxx”, the max
talk time, will be 200*6/1=1200 seconds.
J Note: Tel prefix * is used as a default rate, you need to create it to
work.
Search Condition:
You can search a user by User ID, Prepaid or Postpaid condition as figure
7.7-7.
VIP-2100 User’s manual - 70 -
Figure 7.7-7
Number Replace
The purpose of “Number Replace” is to replace called number or
calling number for PSTN or IP. It must be used in call flow to take effect.
Step 1: It is useful for real PSTN number to virtual VoIP number replacement.
Click Number Replace to add a new Number Replace Group, add as
figure 7.8-1.
Figure 7.8-1
Field Description:
• Group ID: 1 (Number Replace Group identify)
• Description: SIP in
Step 2: Click the New created NR and Detail button to add digits setting as
figure 7.8-2.
Figure 7.8-2
Field Description:
• Original Number: Original number filter
• Target Type: ANI or DNIS
• Target Number: The ANI or DNIS are change to target nubmer
Routing Plan
The purpose of Routing Plan is to select T1/E1 trunk and channels by
your preference when there is a call from IP side to PSTN side. The PSTN
must be used in call flow edit or line hunting component to take effect.
Hunting Group
Start Path: Configuration→Routing Plan→Hunting Group
VIP-2100 User’s manual - 71 -
Figure 7.9-1
Parameters:
• Group ID: Hunting Group ID
• Description: Description of Hunting Group
• Hunting Method: Route selection
o Random: Random select a trunk within this hunting group
o Priority: Select a trunk by priority. (Priority 1 has lowest priority; 9
Parameters:
Group: 4 Description: FET Trunk 02 and 03
• Interface ID: Interface ID
• Trunk ID: trunk id for group 4
• Priority: Trunk priority
• Channel Mask: Channel mask for incoming or outgoing calls (refer
T1/E1 Trunk Configuration)
J Note: When a Route Plan channels mask is cooperated to trunk
channel mask to decide the channel availity 17~31 channels are
available:
Example 1:
Trunk ID: 0 channel mask: 0xffffffff
Route Plan channel mask: 0x0000ffff
Available channel: 0x0000ffff (17~31) channels.
Example 2:
Trunk ID: 0 channel mask: 0xffff0000
Route Plan channel mask: 0xffc00000
Available channel: 0xffc00000 (1~9) channels.
VIP-2100 User’s manual - 72 -
Call Routing
The call routing can be used for hunting a PSTN trunk by prefix.
Start Path: Configuration→Routing Plan→Call Routing
Figure 7.9-3
Parameters:
• Group ID: Select the T1/E1 according to the selection of Hunting Group
ID when dialed number is matched
• Number To Route: The dialed telephone number to be matched
• Matched ANI Prefix: Calling party number used to filter
• Allow Use Others: To select other T1/E1 trunk when all trunk are busy
at your desired Hunting Group.
o Allowed: The call will use other T1/E1 trunks which is not belong to
the selected hunting group
o Forbad: The call will be disconnected immediately
Radius Setting
When you have an external RADIUS server to do the AAA
(Authorization, Authentication and Accounting), enter the correct parameter to
the Radius setting. It must be used in call flow to take effect.
Start Path: Configuration→Radius Setting
VIP-2100 User’s manual - 73 -
Figure 7.10-1
Parameters:
• Auth IP: Radius Authentication Server IP address (default)
• Auth Port: Radius Authentication Server Port
• Acct IP: Radius Account Server IP address
• Acct Port: Radius Account Server Port
• Backup Auth IP: Backup Radius Authentication Server IP address
• Backup Auth Port: Back Radius Authentication Server Port
• Backup Acct IP: Back Radius Account Server IP address
• Backup Acct Port: Back Radius Account Server Port
• Secret Key: The shared secret key with RADIUS Server
• Max Retry: The maximum retry times
• Response Time (sec): The maximum wait for response time from
RADIUS Server
• Auth Retry Interval (sec): The internal to resend the Authentication
packet to RADIUS Server.
• Acc Retry Interval (sec): The internal to resend the Account packet to
RADIUS Server.
• Switch Threshold: Switch to alternate RADIUS Server when failures are
occurred more than switch threshold.
• Auto Inactive: Auto inactive an unused or not
o Disable: Don’t auto inactive
o Prepaid User: Auto inactive prepaid user only
o Postpaid User: Auto inactive postpaid user only
o All User: Auto inactive all unused user
• Inactive prepaid: The minimum credit point threshold for a prepaid user
to be inactived
• Inactive Period: The max unaccess days for a postpaid user to be
inactived
• Charge Method: Billing charge method selection
o Point per Second: Point / calling rate = seconds
o Second per Point: Point * calling rate / charge point = seconds
• Auto Clean: Auto clean the inactive user
VIP-2100 User’s manual - 74 -
o Disable: Don’t auto clean inactive user
o Prepaid User: Auto clean prepaid user only
o Postpaid User: Auto clean postpaid user only
o All User: Auto clean inactive user
• Clean Filter: Auto clean filter
o None: Auto clean users exceed clean period without access the
network
o Inactive: Auto clean only to inactive users
• Clean Period: The maximum unaccess days to clean up. When the
clean filter is set Inactive, the unaccess day is start counting when the
user is inacived
Apply Change
1. Some of modification needs to restart system before it is effective to system
operation. “Apply the change” shows “The change you mode need to
restart the system for apply please confirm to restart or do it later?”
Click on OK button to reboot the system.
Figure 7.11-1
2. For the modification can be changed to fly, “Apply the Change” shows “Are
you sure to apply the running system?” Click on OK button to take
effecting.
Figure 7.11-2
VIP-2100 User’s manual - 75 -
Chapter 8 System Control
System
Start path: Click Control→System
Figure 8.1-1
Parameter:
• Soft Reset: Soft Reset at VIP-2100
• Restart: Restart the VIP-2100
• Shutdown: Shutdown the VIP-2100
System Time
Timezone Setting
Step 1: If you would like to use timezone, click Timezone button to setup the
system timezone as figure 8.2-1.
Figure 8.2-1
Stardand:
Step 2: Select the Standard option to setup the system predefined time zone
as figure 8.2-2
Figure 8.2-2
VIP-2100 User’s manual - 76 -
Parameter:
• Time Zone:
o Standard: Use a predefined standard time zone (Refer Timezone
to Country Mapping List)
o Customize: Use a user defined time zone
• Auto Daylinght Saving: Auto adjust daylinght saving time or not
User defined timezone :
Step 3: Select the Customized option and enter the time zone bias to set a
user defined timezone as figure 8.2-3
Figure 8.2-3
Parameter:
• Daylight Bias: The offset added to the Bias when the time zone is in
daylight saving time
• Daylight Start: The date that a time zone enters daylight time
o Month: 01 to 12
o Week Day: Sunday to Saturday
o Apply Week (Day:01 to 05, Specifies the occurrence of day in the
month; 01 = First occurrence of day, 02 = Second occurrence of
day, ...and 05 = Last occurrence of day)
o Hour: 00 to 23
• Standard Start: The date that a time zone enters daylight time
o Month: 01 to 12
o Week Day: Sunday to Saturday
o Apply Week (Day:01 to 05, Specifies the occurrence of day in the
month; 01 = First occurrence of day, 02 = Second occurrence of
day, ...and 05 = Last occurrence of day)
o Hour: 00 to 23
VIP-2100 User’s manual - 77 -
Network
DNS Server Setting:
Step 1: After successfully logon to the system, we need to change the
network configuration. Click Control→Network to setup the network
parameters as figure 8.3-1.
Figure 8.3-1
Parameter:
• 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
SNMP
Start path: Click Control→SNMP→Community
Figure 8.4-1
Parameter:
VIP-2100 User’s manual - 78 -
• Community Name: Community name for network manager system
accessing
• Access Rights: Giving access right to the community
Start path: Click Control→SNMP→Trap
Figure 8.4-2
Parameter:
• Trap Community: Trap community name for NMS
• Trap Host: Trap host IP address
JNote: It takes around 1-minute to update SNMP configuration and
display successful message.
Prompt Manager
Start path: Click Control→Prompt Manager
Figure 8.4-1
VIP-2100 User’s manual- 79 -
☻Note:
1. You mast has a sound card in your PC to record the voice. You
need to set Network security in order to execute this recording.
Click Tool→Internet Option→Security→Custom Level.
2. Enable the following security to active setting:
Voice prompt editor:
- Download unsigned ActiveX control: Enable
- Initialize and script ActiveX control not marked as safe: Enable
VIP-2100 User’s manual - 80 -
New, Record:
Step 1: Make sure you have installed microphone or other device when you
want to record, Click New and Record buttons to record as figure
8.4-2.
Figure 8.4-2
Stop, Pause, Play:
Step 2: Click Stop or Pause button to stop record, and click Play button to
listen the voice prompt as figure 8.4-3.
Figure 8.4-3
Save:
Step 3: Click Save button to saving the voice file at local path, and the screen
shows Please input the file path and file name!! (i.e.
c:\irene_test.raw) as figure 8.4-4.
Figure 8.4-4
Save Remote File:
Step 4: Click Save Remote File to saving the voice file at VIP-2100, and the
screen shows “please input the file path and file name!!” (i.e.
9999.raw) as figure 8.4-5
VIP-2100 User’s manual - 81 -
Figure8.4-5
☻Note: The file name must be “ .raw” file format.
Open Remote File:
Step 5: Click Open Remote File button to open voice file at VIP-2100 and
screen shows Voice File List as figure 8.4-6.
Figure 8.4-6
Open:
Step 6: Click Open button to open local host voice file and screen shows
Choose File as figure 8.4-7.
Figure 8.4-7
Close:
Step 7: Click Close button to close the voice file as figure 8.4-8.
VIP-2100 User’s manual - 82 -
Figure 8.4-8
Copy:
Step 8: Select the desired voice range and click Copy button as figure 8.4-9
8.4-9
Paste:
Step 9: Click Paste button to paste the voice range as figure 8.4-10.
Figure 8.4-10
Cut:
Step 10: Select the desired voice range and click Cut button as figure 8.4-11.
Figure 8.4-11
Save As: Refer the Section “Save”
VIP-2100 User’s manual - 83 -
Save Remote As: Refer the Section “Save Remote File”
Undo:
Step 13: Click Undo button to return modification, you can see the
configuration that haven’t be changed as figure 8.4-12.
Figure 8.4-12
Redo: Refer Section “Undo”
Zoom Zoom In Zoom Out:
Step 14: Select the desired voice range click Zoom button as figure 8.4-13.
Figure 8.4-13
Step 15: The screen shows the zoom out voice file range as figure 8.4-14.
VIP-2100 User’s manual - 84 -
Figure 8.4-14
Delete Remote file:
Step 16: Click Delete Remote file button to delete remote voice file as figure
8.4-15.
Figure 8.4-15
Call Flow Editor
Please refer section “Call Flow Editor”
Account Manager
Please refer section “Account Manager”
Upgrade
Step 1: Click “Control→Upgrade” to upgrade the software as figure 7.5-1.
☻Note: You can click Clear button to clear all event log.
See the detail event log:
Double click the log or select the log and click detail to see the log
detail.
Figure 9.3-2
Event Description:
Event ID
[GK]: [xxx.xxx.xxx.xxx:xxxx] not found or
8003
8700 VoIP Gateway application on the fly change On the fly change (system change)
8703
9500 Gateway application started VoIP Gateway program start
9500 AAA Mgr application started AAA Manager program start
9500 TelnSvr application started Telnet Server program start
9501 VoIP Board (0) started Interface (ID:0) start
9502 H323 stack started H323 stack start
9503
9504 PSTN trunk (0) alarm clear Connect to PSTN
9505
9600 SNTP client application started Failed / Success to connect SNTP server
registered failure
[SIP Register]: [xxx.xxx.xxx.xxx:xxxx] not found
Registered to H323 Gatekeeper
Registered to SIP Registratar Server
D Channel and Trunk ID (ID:0) available
VIP-2100 User’s manual - 89 -
Debug Info
Start Path: Click “Monitor→Debug Info”
Figure 9.4-1
Filed Description:
• Get Log: Get debug log (-1~999)
• Search: Search debug logs
• Clear: Clear log
Ping
You can use the “Ping” to check an IP is active or not.
Start Path: Configuration→Ping
Figure 9.5-1
Field Description:
• Host IP Address: The IP address to ping
VIP-2100 User’s manual - 90 -
Chapter 10 Telnet & RS-232 Configuration
VIP-2100 also can support to be managed by Telnet or Console port
(RS-232) for basic operations.
Interface:
v Network: TCP/IP Telnet
v RS232:
- Connect using: COM1
- Baud Rate: 9600
- Data bits: 8
- Parity: None
- Stop bits: 1
- Flow Control: None
- Wire: Null modem line (crossed)
Logon VIP-2100 by Telnet
Use Windows build-in Hyper Terminal or other telnet terminal emulator
to login (e.g. telnet 192.168.111.111:10086). User ID & password will be
required for login (default login user id: admin, password: admin & user id:
root, password: root).
Command List:
Command Description
echo Auto echo on or off
eventlog Clean or show system log message
exit Quit the current session
ipconfig Configure or show network information
ping Check an IP address is available or not
reboot Reboot
reset Soft-reset
shutdown Shutdown
time Reset or show system time.
timezone Setup or show system timezone
useradmin Manage user account.
help & ? View command list
[root#]ipconfig Show current network configuration
USE FIXED IP (or DHCP)
IP Address : 192.168.5.113
Subnet Mask : 255.255.0.0
Default Gateway : 192.168.1.254
DNS Servers : 192.168.5.1
168.95.1.1
[root#]ipconfig –delete
dns
Delete the DNS servers setting
USE FIXED IP
IP Address : 192.168.5.113
Subnet Mask : 255.255.0.0
Default Gateway : 192.168.1.254
DNS Servers :
[root#]ipconfig –dchp Enable DHCP
USE DHCP
VIP-2100 User’s manual - 92 -
IP Address : 192.168.5.10
Subnet Mask : 255.255.0.0
Default Gateway : 192.168.1.254
DNS Servers : 192.168.5.1
168.95.1.1
[root#]ipconfig –ip
61.220.126 28 –mask
255.255.0.224 –gateway
61.220.126.1
[root#]ipconfig –ip
61.220.126.115
[root#]ipconfig –dns
210.59.126.53
Use fixed network configuration
USE FIXED IP
IP Address : 61.220.126.28
Subnet Mask : 255.255.255.1
Default Gateway : 61.220.126.254
DNS Servers :
Changes IP address only.
USE FIXED IP
IP Address : 61.220.126.115
Subnet Mask : 255.255.255.1
Default Gateway : 61.220.126.254
DNS Servers :
Changes DNS configuration only.
USE FIXED IP
IP Address : 61.220.126.115
Subnet Mask : 255.255.255.1
Default Gateway : 61.220.126.254
DNS Servers : 210.59.126.53
Ping: Check an IP address is available or not
Command Purpose
[root#] ping ? Usage: ping IP.
Example: ping 127.0.0.1
[root#]ping 61.220.126.1 Ping result
Reply from 61.220.126.1 bytes=64 time=1ms TTL=29
Reply from 61.220.126.1 bytes=64 time=1ms TTL=29
Reply from 61.220.126.1 bytes=64 time=1ms TTL=29
Reply from 61.220.126.1 bytes=64 time=1ms TTL=29
Reboot:
Command Purpose
[root#] reboot ? Reboot System
Are You Sure? (Y/N)
[root#]reboot
Are You Sure?(Y/N)y
VIP-2100 are rebooting
Shutdown:
Command Purpose
[root#] shutdown ? Shutdown System
Are You Sure? (Y/N)
[root#]shutdown
Are You Sure?(Y/N)y
VIP-2100 are shutting down
Reset:
Command Purpose
[root#] reset ? Soft reset System
Are You Sure? (Y/N)
[root#]reset
Are You Sure?(Y/N)y
VIP-2100 User’s manual - 93 -
Time: Reset or show system time
Command Purpose
[root#] time ?
[root#]time
[root#]time 2003-07-29
Usage : time YYYY-MM-DD HH:NN:SS
Example : Time 2002-01-01 12:00:00
Show current time
The current time is 2003-06-20 15:17:30
Change system bios time
23:14:53
Timezone: Setup or show system timzone
Command Purpose
[root#] timezone ?
Fixed Zone List:
01. Afghanistan Standard Time
03. Arab Standard Time
05. Arabic Standard Time
07. AUS Central Standard Time
09. Azores Standard Time
11. Cape Verde Standard Time
13. Cen. Australia Standard Time
15. Central Asia Standard Time
17. Central European Standard
Time
19. Central Standard Time
21. Dateline Standard Time
23. E. Australia Standard Time
25. E. South America Standard
Time
27. Egypt Standard Time
29. Fiji Standard Time
31. GMT Standard Time
33. Greenwich Standard Time
35. Hawaiian Standard Time
37. Iran Standard Time
39. Korea Standard Time
41. Mexico Standard Time 2
43. Mountain Standard Time
45. N. Central Asia Standard
Time
47. New Zealand Standard Time
49. North Asia East Standard
Time
51. Pacific SA Standard Time
53. Romance Standard Time
55. SA Eastern Standard Time
57. SA Western Standard Time
59. SE Asia Standard Time
61. South Africa Standard Time
63. Taipei Standard Time
65. Tokyo Standard Time
67. US Eastern Standard Time
69. Vladivostok Standard Time
71. W. Central Africa Standard
Time
73. West Asia Standard Time
75. Yakutsk Standard Time
02. Alaskan Standard Time
04. Arabian Standard Time
06. Atlantic Standard Time
08. AUS Eastern Standard Time
10. Canada Central Standard
Time
12. Caucasus Standard Time
14. Central America Standard
Time
16. Central Europe Standard
Time
18. Central Pacific Standard
Time
20. China Standard Time
22. E. Africa Standard Time
24. E. Europe Standard Time
26. Eastern Standard Time
28. Ekaterinburg Standard Time
30. FLE Standard Time
32. Greenland Standard Time
34. GTB Standard Time
36. India Standard Time
38. Israel Standard Time
40. Mexico Standard Time
42. Mid-Atlantic Standard Time
44. Myanmar Standard Time
46. Nepal Standard Time
48. Newfoundland Standard
Time
50. North Asia Standard Time
52. Pacific Standard Time
54. Russian Standard Time
56. SA Pacific Standard Time
58. Samoa Standard Time
60. Singapore Standard Time
62. Sri Lanka Standard Time
64. Tasmania Standard Time
66. Tonga Standard Time
68. US Mountain Standard Time
70. W. Australia Standard Time
72. W. Europe Standard Time
74. West Pacific Standard Time
VIP-2100 User’s manual - 94 -
[root#]timezone
Usage1 : timezone Zone (1 to 75) AutoDaylight (Y or N)
Example1 : timezone 1 Y
Usage2 : timezone -custom Bias DaylightBias DaylightStart
StandardStart
Bias : -12:00 to +13:00
DaylightBias : -12:00 to +13:00
DaylightStart :
MM (Month: 01 to 12) ;
WD (Day of week: 00 to 06)
DD (Day:01 to 05 ;Specifies the occurrence of day in the
month;
01 = First occurrence of day,
02 = Second occurrence of day, ..., 05 = Last occurrence of
day
HH (Hour:00 to 23)
StandardStart :
MM (Month: 01 to 12) ;
WD (Day of week: 00 to 06)
DD (Day:01 to 05 ;Specifies the occurrence of day in the
month;
01 = First occurrence of day,
02 = Second occurrence of day, ..., 05 = Last occurrence of
day
HH (Hour:00 to 23)
Example2 : timezone -custom +08:00 -01:00 04-00-01-02 10-0005-02
Show current timezone info
Time Zone : (40) Mexico Standard Time (GMT -06:00)
Daylight Bias : -01:00
Daylight Start : 05-00-01 02:00
Standard Start : 09-00-05 02:00
Auto Daylight : Y
[root#]timezone 40 n Use pre-defined timezone
Time Zone : (40) Mexico Standard Time (GMT -06:00)
Daylight Bias : -01:00
Daylight Start : 05-00-01 02:00
Standard Start : 09-00-05 02:00
Auto Daylight : n