CHAPTER 8 SYSTEM CONTROL...................................................................................................87
SYSTEM ..............................................................................................................................................87
SYSTEM TIME .....................................................................................................................................87
A Universal VOIP/SS7 Gateway – Enabling SS7 to VoIP Network
Wellgate 5290 is a universal VOIP/SS7 GATEWAY which navigates the calls in
between H.323, SIP and PSTN freely, not simple PSTN to VOIP calls or vice verse.
It can be easily used to implement SIP and PSTN, H.323 and PSTN, SIP and H.323,
SIP and SIP, H.323 and H.323 calls simultaneously. With built-in SS7/ISUP interface,
Wellgate 5290 bridges traditional PSTN/SS7 to VOIP network. Also it provides
PSTN/VOIP IVR helps service provider to create their own voice service very quick.
Wellgate 5290 provides an easy to use web-based managing interface. Administrator
can use the web based drag and drop call-flow editor to design their VOIP service
very quick and without losing any flexibilities. A web-based voice prompt
management GUI is also provided to simplify the deployment of IVR related service.
Key Features
Navigate Call Freely in SIP, H.323 and PSTN
Support SIP RFC 3261 and ITU-T H.323 V5 Simultaneously
Up to 4 Programmable E1/T1 Trunks
PSTN Signaling: ISDN/PRI, CAS, MFC R2, QSIG, SS7/ISUP
Support with Voice or Signal only SS7 Link
Support Audio Codec G.711, G.723.1, G.729A, GSM
SIP to PSTN Call and vice versa
H.323 to PSTN Call and vice versa
H.323 to SIP Call and vice versa
SIP to SIP Call
H.323 to H.323 Call
Built-in Universal VOIP Address Book
Support up-to 16 Multiple SIP Proxy Servers
Support SIP Proxy, Gatekeeper and P2P Calls Simultaneously
Support Early Media and SIP Delay Media
Support RADIUS Authentication, Authorization and Accounting
Intelligent PSTN Call Routing and in-Trunk Hunting
Support Flexible VOIP Routing and Account Code
Flexible Digit Manipulation Plan
Support Calling/Called Number Replacement
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Technical Specification
Interface
- Two 10/100MB Ethernet Ports (Host & VoIP stream)
- E1 SS7 Interface
- Up to 4 T1/E1 (120 Ohm-RJ48C connectors)
75 Ohm needs external 3rd party BNC/RJ-48C adapter cables
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
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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
T1 CAS DTMF/R1MF
7
T1 ISDN PRI Support
8
9 E1 SS7 Support ISUP with voice/without voice
Trunk Spans
11
Default Trunk Channel Mask
12
PSTN Line Hunting
13
PSTN Line Hunting Channel Selection
14
15 Overlapping Sending for ISDN incoming call Yes
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
6 H.245 Tunneling
7 Early H.245
8 Cause Code Mapping
SIP Protocol Support
1 Cause Code Mapping
2 HTTP Digest Authentication
3 SIP Call on Hold
4 SIP Early Media/Delay Media
5 SIP Overload Redirect
6 SIP Transfer (unattend)
7 SIP Transfer (attend)
8 SIP/TCP
9 SIP/UDP
4 H.323 to PSTN Call
5 H.323 to SIP Call
6 H.323 to SIP FAX Relay
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
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, Selected disconnect
cause code & behavior
3 RADIUS Accounting
4 Redundant RADIUS Server Support
5 PSTN Prepaid Support
6 VoIP Prepaid Support
Embedded AAA
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 Phone Book Entries
6 Max Call Flow Component
7 Max CDR Keep Days
8 Max Voice File Storage
Manual
1 English User Guide
Yes
Yes, Active/Standby/Auto Failover
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
4096
2048
4096
4096
10000
256
5
10 hours
Yes
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Welltech Computer Co., Ltd.
WellGate 5290 Appearance Description
WellGate 5290 Front Panel:
1
2
Functions:
1: Power LED
2: H/D LCD
3: System Status LED
4: LAN or Management Interface
5: WG5290 Service Interface
6: Power Switch
7: LCD Panel
8: LCD Touch Panel
9: USB
l
WellGate 5290 Rear Panel:
3
4
7
5
Figure 1.1-1
6
8
9
1
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2
3
12
7
Figure 1.1-2
13
11
10
4
8
5
6
9
Welltech Computer Co., Ltd.
Functions:
1: Electric Fan
2: AC Power outlet
3: AC Power switch
4: Keyboard/Mouse
5: Management Ethernet port
6: Service Ethernet port
7: VOIP Ethernet port
8: External Monitor Connector
9: Console port
10: USB (not used)
11: E1 Trunk port
12: SS7 E1 (if you are using SS7 with voice E1, you need to configure the corresponding
SS7 settings according to Dialogic document and connect 2
order to separate voice from SS7 E1 to voice E1)
nd
E1 SS7 E1 to Voice E1 port in
13: Voice E1 derived from SS7 E1. It is used only for SS7 with voice E1 and
will be connect to E1 Trunk Port (10).
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Welltech Computer Co., Ltd.
Chapter 2 Logon WellGate 5290
After connected E1/T1 & Ethernet cables into the WellGate 5290,
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 WellGate 5290, you need to
have Java Standard Runtime (1_4_1_02) to make it work. Please refer to
Appendix 2 Java plug-in Installfor detail.
Logon WellGate 5290
Setp1: Start IE5.0 (or later version) to navigate WellGate 5290 Management
System by typing the default IP addresses (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 255.255.255.0
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
Step 3: The screen shows the Home Page of WellGate 5290 as figure 2.1-3.
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Welltech Computer Co., Ltd.
Figure 2.1-3
Figure 2.1-4(SS7)
Network Configuration
Step 1: After successfully logon to the system, we need to change the
network configuration. Click Control > Network -> SIP Service
Interface to setup the network parameters as figure 2.2-1.
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Figure 2.2-1
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
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☻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
time zone 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.
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.
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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 as you need. Apply the change as
figure 2.4-2.
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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”
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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 WellGate 5290 as
figure 2.5-2.
Figure 2.5-2
Step 3: The screen shows the Home Page of WellGate 5290 as figure 2-5-3.
provides the Dialogic SS7 configuration utility to edit the Dialogic SS7 board
configuration file (config.txt). After you complete the editing of config.txt, you can
upload to the system and soft-reset to take effective. However, some advance
configuration (e.g. SS7 with voice) is not supported by the Dialogic utility. User need
to use the normal editor to edit it.
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Frequency used parameters:
• SS7: Enable
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: 222.173.67.138
• E.164 Tel: 177
• Register H.323 ID: Welltech
Step 3: You can see the screen display the new configuration of the H.323
Configuration as figure 3.3-3.
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Welltech Computer Co., Ltd.
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.
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Welltech Computer Co., Ltd.
Figure 3.4-1
Select SIP Proxy ID 0 and click modify as follows
Frequency used parameters:
• Active Mode: Active
• Register Server: SIP Proxy Server Register Server
• Register Port: SIP Proxy Server Register Port
• Register User: SIP User to register
• Register Password: SIP User Password to register
• Register TTL: 300 (time to live)
Step 2: You can see the screen display the new configuration of the SIP
Configuration as figure 3.4-2.
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Welltech Computer Co., Ltd.
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
Step 2: Enter the related parameters and click Apply button as figure 3.5-2.
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Welltech Computer Co., Ltd.
Figure 3.5-2
Field Description:
• Group ID: 0 (DM Group identify)
• Description: H.323: H.323 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
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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-add002 (belong to this DM group)
• Insert: The Insert digits
H.323 incoming call
↓
Dialed number: 582265699
↓
Match the pattern 5
↓
Add 002 (Insert)
↓
New dialed number becomes 002582265699
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: 2-sophie (belong to this DM group)
• Drop: 113 (drop digits)
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Welltech Computer Co., Ltd.
SIP incoming call
↓
Dialed number: 11307688222
↓
Match the pattern 113
↓
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: 02822656 (pattern to be matched)
• Group ID: 3-PSTN (belong to this group id)
• Drop: 02822656 (drop digits)
PSTN incoming call (DNIS mode)
↓
Dialed number: 028226561001
↓
Match the pattern 02822656
↓
Delete 02822656 (Drop)
↓
New dialed number becomes 1001
☻Note: Digit Manipulation have to tapped for PSTN Side
(Trunk>Outbound/Inbound DM Group), VoIP Side
(VoIP>Outbound/Inbound DM Group) or Call Flow (refer to
section “Call Flow Editor”) to take effect.
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Welltech Computer Co., Ltd.
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
•
•
•
•
• 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
Load Call Flow: Load call flow from WellGate 5290
Save: Save a call flow in WellGate 5290
Cut: Cut a component
Copy: Copy a component
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Welltech Computer Co., Ltd.
•Show Component Table: Show all component table
Step 2: Drag and prop the required component icon into the workspace as
figure 4.2-1
Figure 4.2-1
Right click the component to bring up the component propriety to
setup parameter:
•AAA: Send Authorization or Authentication for validation
Figure 4.2-2
o Type: AAA type selection
- Authorization: Send RADIUS Authorization packet out
- Authentication: Send RADIUS Authentication packet out
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Success To: Success to component
Failed other to: Failed to component
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
☺
Note: Detail response attributes, please refer RADIUS Format
Attributes
•
Answer: Answer incoming call (PSTN only)
•Branch: Play an announcement and branch into different route
Figure 4.2-3
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
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Welltech Computer Co., Ltd.
Figure 4.2-4
•CDV: Collected Digit Validation
Figure 4.2-5
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 Invalid To: If the checked variable is not success to validate
•
CIB: Call Information Branch
Figure 4.2-6
o Info Type: Information type selection
- ANI: Calling Number
- DNIS: Called Number
- IP: IP Address or network (e.g. 192.168.0.0)
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- 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 info type to be validation
-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
Figure 4.2-7
Figure 4.2-8
•
WellGate 5290 User Guide – V6.1 - 38 -
Cut Rule: Cut a system variable into different parts
Welltech Computer Co., Ltd.
Figure 4.2-9
o Cut From: Cut start digit from (start from 1)
o Cut To: Cut end digit to
o Assign To: Store the cut result into
• Disconnect: Disconnect the call
• DM: Digit Manipulation
Figure 4.2-10
o DM Parameter: Manipulation ANI or DNIS
o DM Group ID: Apply to DM group
• Make Call: Make Call to PSTN or H.323/SIP site
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Welltech Computer Co., Ltd.
Figure 4.2-11
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 WG5290 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
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
WellGate 5290 User Guide – V6.1 - 40 -
-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
Welltech Computer Co., Ltd.
Figure 4.2-12
•PB: PlayBalance for prepaid purpose
Figure 4.2-13
o Voice File: Voice prompt file
o Language: Play balance language section
-English
-Chinese.
o Interrupted: Voice can be interrupted or not
•PCUI: Prompt and Connect User Information
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Welltech Computer Co., Ltd.
Figure 4.2-14
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
Figure 4.2-15
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
Figure 4.2-16
o Success To: If fine an available channel by system setup (call
hunting)
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o Failed To: If not fine an available channel by system setup (call
hunting)
•Set Data: Assign value to a variable
Figure 4.2-17
o Assign To: Assigned variable
o Use SysParam: Use system parameter to replace or not
o Value: ANI, DNIS, User ID or other digits. You can also use
database number to generate the required random number.
o Support random ANI (use ? to indicate random digit)
Configuration Management provides a way to save and reload the
system configuration for future use.
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Figure 4.4-1
Backup the working configurations:
Step 1: 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
4.4-2.
Figure 4.4-2
Restore configuration:
Step 2: 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 4.4-3.
Figure 4.4-3
Compact the database file:
Step 3: In order to optimize the system performance, you can optional
compact the database by click Compact button as figure 4.4-4.
Figure 4.4-4
☺
Note: Please make sure that there is no others person to use database
concurrently.
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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 system or
☺Digit Manipulation: Please refer to section “Digit Manipulation”
Network Configuration
Please refer to section “Network Configuration”
Account Manager
Please refer to section “Account Manager”
Interface Configuration
Please refer to section “Interface Configuration”
H.323 Configuration
Step 1: Change Register to Gatekeeper to “No” to enable peer to peer mode
as figure 5.1-1.
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Frequency used parameters:
• Register to Gatekeeper: No
Figure 5.1-1
VOIP Hunting
For making a Peer-to-Peer call, the IP device must has an address
record in the phone book for routing.
Step 1: To add a new VOIP routing group for the peer to peer calls, click
VOIP Hunting -> New to add as figure 5.2-1.
Figure 5.2-1
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Field Description:
• Group ID: The Group ID
• Description: The description of this VOIP hunting group ID
Step 2: To add a routing point into the VOIP hunting group, click the hunting
group ID and Detail button as figure 5.2-2.
Figure 5.2-2
Field Description:
• Name: The used name
• Tel/Prefix: The matched prefix
• Trans Address: transport address
- H.323 Call: 192.168.5.102 or 192.168.5.102:1720
- SIP Call: sip:1001@192.168.5.61 or sip:8001@192.168.5.61:5060 or
sip:8001@ctivnet.net
• Use SIP Proxy: use the selected SIP proxy instead of Transport
address
• Priority: It is used for priority hunting. Higher number is indicating the
high priority to be used. Same priority routing item will be round-robin
hunted.
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• Remove Prefix: Remove the TEL/Prefix from the final destination
number (DNIS) or not
• Add Prefix: After removed the prefix (if remove prefix is checked), add
the “Add Prefix” in the front of final destination number (DNIS)
• DNIS DM Group: To execute the DNIS DM Group or not. It will be
executed after “Add Prefix”.
• Working Time: The effective time for this route. It needs to be used
with the working day together.
• Working Day: The effective week day for this route. It needs to be used
together with working time.
• Allow no ANI: Does this route is allowed for a call without ANI or not.
Step 3: The hunting can be stopped based on the following reason code. It
can be used as a load balancing purpose.
Figure 5.2-3
Digit Manipulation
Please refer to section “Digit Manipulation”
Call Flow Editor
Please refer to section “Call Flow Editor”
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Call Flow (P2P Mode):
to 1009
H.323 in
Welltech Computer Co., Ltd.
Start:1000
CTB:1001
PST N to:1011
H323 to: 1009
SIP to: 1008
to 1008
SIP in
1011
PSTN in
DM: 1013
Call to PSTN
DM: 1009
Call to PSTN
Make Call: 1007
Call to PSTN
Finish to
Failed other to
DM: 1008
Call to PSTN
Disc: 1005
Disconnect
Make Call: 1004
Make
Peer to Peer
call to SIP
Success /Failed to
Quit: 1006
Disconnect
Configuration Manager
Please refer to section “Configuration Manger”
Apply Change
Please refer to section “Apply Change”
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S
SIP Phone
Chapter 6 SIP to H.323 Mode Configuration
Environment used in this chapter
Process:
SIP>H.323 Call: DNIS (8861001)>DM (SIP In Drop)>Make H.323 (1001)
• 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 Keep days: CDR system keeping days
• Hot Swappable: Hot swappable support (reserved)
• First Digit Timeout: The max time (in second) waits for receiving the first
digit entered (5~20 sec).
• Inter Digit Timeout: The max time (in second) waits for the between two
digits (5~20 sec).
• Debug Level:
o Critical: Show critical error messages only
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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
• SS7 Debug Level: SS7 related debug info or not.
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 hit.
• First Response Timeout: The default maximum time to wait for response.
It’s depended on the network speed.
• No Answer Timeout: The default maximum time (in second) to wait the
remote party Answer (pick up phone).
• Ext. AAA Failure Opt: Bypass or disconnect incoming calls when external
RADIUS cannot be reached.
• Version: 6.0
• Data Base Number Select Mode: The Number Select Mode In Data Base
• Enable System Log: Enable to send system information to syslogD
Server or not
• SyslogD Server IP 1, 2: syslogd server IP address
• 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 Internet work 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
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• PCM Idle Pattern: This pattern will be sending on each B channel PCM
time slot when the channel is idling (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
voice quality. Set to 12 will have long voice delay but with better voice
quality
• EC Tail Length: Echo Cancellation Length, default value is 32ms
• 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
• Detect Fax Mode: T.38 fax detecting method
o Answering Fax only: Detect answer tone to initiate T.38
o Fast Fax Detect: Detect calling and answer tone for T.38. Use
Answering fax only when fast fax detect miss detect a fax call.
• Auto Clock: Automatic detect trunk clock source. It is recommended to
specify clock source instead of use auto clock.
• Local Reference: TDM bus reference clock. Please set to a connected
trunk.
• Overlap Digit: Enable to receive ISDN/PRI Q.931 overlapping sending
or not.
• DTMF Interval: Wait overlapping digits time out
• CAS Send Restart: When channel is blocked, send restart or not.
• Active T38 when Open Channel: Re-open DSP channel when T.38 is
started.
• PSTN Show Display Name: Send ISDN display name IE or not
(network side only)
• SIP Show Display Name: Relay ISN display name to SIP or not
• Reset Trunk When Link Up: When trunk is started, restart channels or
not.
Dial Plan Configuration
Dial Plan can be used to assign the ISDN number plan based on prefix
setting.
Start Path: Configuration>Interface>Dial Plan
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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
• Apply To: Trunks apply to
Overlap Configuration
It is used for incoming ISDN dialing plan.
Start Path: Configuration>Interface> Overlap
Figure 7.4-1
Click New to create a overlap dialing plan as follows:
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Figure 7.4-2
• Prefix: Matching prefix (longest match first)
• Max Length: Max length for this prefix dial plan. The dial plan will be fulfill
if max length is met.
• Apply To: Apply Target to which trunk or all trunk
T1/E1 Trunk Configuration
Start Path: Configuration>Interface>Trunk
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Figure 7.5-1
Basic Parameter Description:
• Interface ID: System parameter
• Trunk ID: System parameter
• Trunk Type: T1or E1 selection (SS7 support E1 only)
• Description: Description for this trunk ID
• Termin Side: Network site or User Site (normally, you set to “user side”
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 VOIP incoming only: Allow the VOIP 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
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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:
oFor 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
oFor 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
oFor T1:
- T1 CAS
- T1 RAW CAS
- T1 NI2 ISDN
- T1 4ESS ISDN
- T1 5ESS 9 ISDN
- T1 5ESS 10 ISDN
- 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
oFor 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
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• 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)
• Clock Master: PSTN trunk clock source
• Framing Method: Automatic framing
• SS7 Circuit Group Mode: Enable the this SS7 Circuit group (IN Service)
or not (Out Service)
• SS7 Group ID: Circuit Group ID in SS7 config.txt (refer to
ISUP_CFG_CCTGRP gid)
• SS7 Circuit Identifier 1: First CID in SS7 config.txt (refer to
ISUP_CFG_CCTGRP base_cid)
• SS7 Circuit Identifier Count 1: set to 15 (internal user only)
• SS7 Circuit Identifier 2: Start CID for time slot 17 (refer to
ISUP_CFG_CCTGRP based_cid + offset)
• SS7 Circuit Identifier Count 2: set to 15 (internal use only)
• Input Gain: Voice Gain from PSTN to IP side (default: 0 db)
• Output Gain: Voice Gain from IP to PSTN side (default: 0 db)
• Q.931 General Opt.: used for Q.931 general behavior.
o0x00000001: No sending of STATUS message on receipt of an
unknown IE. If this bit is set, then ISDN Stack does not generate a
STATUS message after having received a message containing 1 or
more unknown/unrecognized IE(s). This bit applies only for network
variants for which the sending of STATUS in this case is optional
(where recommendation says “ a SATUS MAY be returned”)
0x00000002: No sending of STATUS message on receipt of an
o
optional IE with invalid content. If this bit is set, then ISDN Stack
does not generate a STATUS message after having received a
message containing 1 or more optional IE(s) with invalid content.
This bit applies only for network variants for which the sending of
STATUS in this case is optional (where recommendation says “a
STATUS MAY be returned”).
o
0x00000004: Unknown/unrecognized Facility IE accepted. If this bit
is set, then NS does not reject a received message containing a
Facility IE that it does not recognize and acts as if the IE was
correct. It applies in network variants where a complete ASN1
decoding is made on Facility IE
o0x00000080: Only applicable when configuration is Euro ISDN,
User-side, outgoing call . If this bit is set, then the board sends a
CONNECT_ACK message when receives a CONNECT.
o0x00000100: Only applicable when configuration is ETSI. If this bit
is set, BC and HLC IE’s in Alert, CALL_PROC, CONNECT and
PROGRESS message are rejected.
0x00000200: Only applicable when configuration is 4ESS, DMS,
o
5ESS, N12 or HKT, NFAS-both incoming and outgoing calls. If this
bit is set, then ISDN calls on NFAS have an interface ID different to
the default one (refer to NFAS Appendix). The interface ID value is
taken from the Board param of the NFAS Member Trunk config.
(i.e.,board_params. Trunk Config[i]. Protocol Specific. ISDN Trunk.
ISDN_NFAS_Interface ID).
o
0x00000800: Only applicable when configuration is 4ESS, DMS,
5ESS or N12. If this bit is set, the channel ID is always set to
EXPLICIT interface ID, even if the B-channel is on the link bearing
the D-channel.
o0x00008000: Only applicable when configuration is ETSI. If this bit
is set, Mu-law (μ-Law) is also accepted. Otherwise, only A-law is
accepted.
o0x00010000: Only applicable when configuration is ETSI. If this bit
is set, calling party number octet 3a is always present, even if
presentation and screening have default values. Otherwise, usual
rule is used.
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o0x00020000: Only applicable when configuration is N12, QSIG or
ETSI. If this bit is set, the call is cleared on reception of STATUS
with incompatible state. Otherwise, no action is taken
• Q.931 Incoming Opt.: used for Q.931 incoming call behavior
o0x00000010: Automatic answering on TELEPHONY incoming
calls︰ When this bit is set, Board sends a CONNECT (answer)
message on TELEPHONY incoming calls.
o0x00000020: Automatic answering on NOT TELEPHONY incoming
calls: When this bit is set, Board sends a CONNECT (answer)
message on NOT TELEPHONY incoming calls.
o0x00000800: When this bit is set, the ISDN Stack forces the
sending of the Channel-id Information Element in the first response
to an incoming call indication. When this bit is not set, The ISDN
Stack forces it only if it is mandatory for the running operator, or if
on NT-side. This have been identified as solving a problem with
Coral PBX that requests a Channel in the first respond to an
incoming call from it .
o0x00001000: When this bit is set, the ISDN Stack does NOT
automatically generate a Setup Ack. When this bit is not set, the
ISDN Stack sends a Setup Ack when it detects overlap receiving in
progress.
o0x00002000: When this bit is set, the ISDN Stack forces the
sending of Channel-id Information Element(IE) in the CallProceeding message. When this bit is not set, the ISDN Stack
forces it only if it is mandatory for the running operator, or if on NTside.
o0x00010000: When this bit is set, the board automatically sends a
Progress-Indicator #8 (“in-band tones/announcements available”) in
the Setup-Ack message that is sent back automatically by the board
when it receives and accepts a Setup message in overlap-receiving
mode (e.g no digit in the Setup), for Voice calls only (BC encoded
as Speech or 3.1-Audio). This enables tone generation to the
Terminals after handset off-hook.
• Q.931 Outgoing Opt.: used for Q.931 outgoing behavior
o0x00000002: Automatic generation of ‘Sending-complete’
information element in SETUP (as result of Place Call).When this
bit is set, Board does not generate automatically the 'Sending-
complete' information element in the SETUP. The User application
must request it in the Place Call function. Otherwise (if this bit is not
set), the Board generates it automatically in the SETUP message
only.
o0x00000010: 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
o0x00000080: When this bit is set, ISDN stack uses the KEYPAD IE
to store the called number digits instead of the CALLED_NB IE.
Only applicable for KOR variant (Korean network). Useful for
Korean switches that do not accept the Called_nb IE.
o0x00000100: When this bit is set, ISDN Stack forces the sending of
a Channel-id IE in an outgoing SETUP message even if it is not
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required by the standard(i.e., it is optional only ), and no Channel-id
was specified in the establishment request. This is useful to
increase compatibility with switches that require it. On PRI lines it
indicates a not yet used channel id, preferred only.
o0x00000200: When this bit is set, ISDN Stack sends G..711 A-Law
in outgoing voice calls. When this bit is cleared, ISDN Stack sends
default G.711-Mu-Law in outgoing voice calls. Applicable for 5ESS
variant(T1).
• Trans Cap: Transfer Capability
o Voice Service
o Data Service
o Modem Service
• Call ID Transfer Type: Call ID transfer type
o Disable Caller ID: default parameter
o Transparent Caller ID
o Relay Caller ID
o Bypass Caller ID
Reset Configuration
Reset a specified channel or full trunk.
Start Path: Configuration>Interface>Detail>Reset
Figure 7.6-1
Start Path: Configuration>Interface>Detail>Reset
Figure 7.6-2
Basic Parameter Description:
• Trunk: Reset trunk ID
• Channel: Rest channel selection
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- All Channel: Reset all channel
- 0~31: Reset 0~30 logical channel to reset
H.323 Configuration
Start Path: Configuration>H.323
Figure 7.7-1
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 you
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 you
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 you expect the SIP Stack to handle simultaneously.
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• 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 form 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.
• Application Pool Page Size: The size of page in the application pool
• Retransmission T1: 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: 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 invite Linger Timer
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 general Linger Timer milliseconds.
(Default Value: 32000)
• Provisional Timer: When a client receives a provisional response, it
continues to retransmit the request, but with an interval of provisional
Timer milliseconds.
• Cancel General No Response Timer: When sending a CANCEL
request on a General transaction, the User Agent waits cancel General
No Response Timer milliseconds before timeout termination if there is
no response for the cancelled transaction.
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• Cancel Invite No Response Timer: When sending a CANCEL request
on an Invite transaction, the User Agent waits cancel Invite No
Response Timer 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 general Request Timeout Timer
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
• Auto Send 183: WellGate5290 always send Call Progress Indicator
(SIP 183) to VoIP party. It can be used for CAS protocol to enable early
media.
• Overload Redirect: SIP overload redirect when WellGate5290 is not
able for service the call
• Send 487 When Recv CANCEL: When receive CANCEL form remote
site, send “487 Request canceled” or not
• Caller ID Mode:
o Local: use WellGate 5290 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 holds tone locally.
• On Hold music: Hold tone music file name
• SIP UnRegist When Trunk Down: Enable or disable un-register when
all trunks are down. It can be used for load balancing.
• SIP Over Load Reason: When all trunk are busy, the reason to send to
caller. Default is 438. When work with WS6500, 438 is required for load
balancing hunting.
• Refer Call Use Refer By User: When a call is been transferred, new
call will use REFER-BY ID instead of original caller ID.
SIP Proxy Server Setting:
WellGate 5290 can support up-to 16 SIP proxy server. You can select the SIP
proxy in call flow editor or VOIP Routing list.
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Figure 7.8-3
• ID: SIP Proxy Identification
• Active Mode: Active or inactive
• Register Server: SIP register proxy server IP Address
• Register Port: SIP register proxy server port number
• Register User: SIP register proxy server User ID
• Register Password: SIP register proxy server User Password
• Register TTL: The registration maximum time to live setting when
registered to the SIP proxy server
• Register Doman Name: SIP register proxy domain
• Outbound Proxy Server: The IP address of the outbound Proxy
• Outbound Proxy Port: The port of the outbound Proxy
• Outbound Proxy User: The User ID of the outbound Proxy
• Outbound Proxy Password: The password of the outbound Proxy
SS7 Configuration
WellGate 5290 is using Dialogic SS7 board with ISUP module. The user
provides the Dialogic SS7 configuration utility to edit the Dialogic SS7 board
configuration file (config.txt). After you complete the editing of config.txt, you can
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upload to the system and soft-reset to take effective. However, some advance
configuration (e.g. SS7 with voice) is not supported by the Dialogic utility. User need
to use the normal editor to edit it.
Start Path: Configuration→SS7
Figure 7.9-1
• SS7: Enable SS7 or not
• Total Channel: read only parameters
• ISUP Module ID (HEX): ISUP message module ID used for Dialogic
SS7 card and configuration (advance user only).
• Tool: Get Dialogic SS7 configuration tool
• Get Config: Download SS7 configuration files from 5290 for Dialogic
SS7 board
• Upload: Upload the modified SS7 configuration files to 5290 for
Dialogic SS7 board
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 (IP Access Control List)
Start Path: Configuration>Access Control>IP ACL
Create a new IP ACL group used for call flow editor by click the New
button.
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Figure 7.10-1
Group ID: IP ACL grouping ID
Description: Description of IP ACL
IP ACL Detail
After create an IP ACL group, you can click the Detail button to add the
corresponding IP list.
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Figure 7.10-2
• 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.
• Account Code: Account Code is part of called number prefix. It can be
used for billing purpose. Account Code can be retrieved by call flow
editor and make it for RADIUS billing user.
• Customer Name: Name of this item
• Default: If there is no Account prefix code can match, this default item
(account code) will be used
• Remove Account Code: Remove Account Code from the DNIS (called
number)
• ANI ACL: If it is enabled, only ANI is matched the ANI ACL group can
be allowed to use.
• Add Prefix: Add the prefix to the DNIS (called number)
• Max concurrent call: The maximum concurrent calls can be allowed for
this IP item.
☻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
Calling Number Screening
Start Path:Configuration>Access Control>ANI ACL
Click New to create an ANI ACL list group
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Figure 7.10-3
• Group ID: ANI ACL Group
• Description: Description of this ANI ACL
Click Detail to add the ANI item into the ANI ACL group.
Figure 7.10-4
Parameters:
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• 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.
DNIS ACL
Start Path: Configuration>Access Control>DNIS ACL
Click New to add a DNIS Access Control List Group.
Figure 7.10-5
• Group ID: DNIS ACL Group ID
• Description: Description
Click Detail to add DNIS prefix into the IP ACL Group.
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Figure 7.10-6
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.
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.
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Figure 7.11-1
Field Description:
• Group ID: 1 (Number Replace Group identify)
• Description: Description
Step 2: Click the New created NR and Detail button to add digits setting as
figure 7.8-2.
Figure 7.11-2
Field Description:
• Original Number: Original number filter
• Target Type: ANI or DNIS
• Target Number: The ANI or DNIS are change to target number
Data Base Number
The purpose of Data Base Number is to change called number for
PSTN or IP. It must be used in call flow to take effect.
Start Path:Configuration>Data Base Number
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Figure 7.12-1
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
Figure 7.13-1
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