Welltech SIPPBX-6200A User Manual

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Version: SIPPBXUM.101

CH1. Overview..................................................................................- 5 -

1.1 Specifications ............................................................................................. - 5 -

1.2 Hardware Overview.................................................................................. - 8 -

1.2.1 The Front Panel ...................................................................................... - 8 -

1.2.2 The Back Panel....................................................................................... - 8 -

CH2. Start to configure SIPPBX 6200A................................- 10 -

2.1 Connection SIPPBX 6200A .................................................................. - 10 -

2.1.1 Network Configuration.........................................................................- 11 -

2.1.2 Extension Configuration ..................................................................... - 14 -

2.1.3 Trunk Configuration............................................................................. - 19 -

CH3. Full Web Configurations ..................................................- 23 -

3.1 Configuration............................................................................................ - 25 -

3.1.1 IP PBX .................................................................................................... - 25 -

3.1.2 Office Call Rule ..................................................................................... - 30 -

3.1.3 Feature Code ........................................................................................ - 33 -

3.1.4 Extension............................................................................................... - 37 -

3.1.5 Auto COnfig (IPv4) .............................................................................. - 45 -

3.1.6 Trunk ...................................................................................................... - 51 -

3.1.7 SIP Trunk Reg....................................................................................... - 56 -

3.1.8 Outgoing Routing................................................................................. - 59 -

3.1.9 Incoming Call Rule............................................................................... - 64 -

3.1.10 Dial Group.......................................................................................... - 67 -

3.1.11 Speed Dial ......................................................................................... - 69 -

3.1.12 Broadcast........................................................................................... - 71 -

3.1.13 Meetme Conf..................................................................................... - 73 -

3.1.14 T.38 FAX ............................................................................................. - 75 -

3.2 Information............................................................................................... - 76 -

3.2.1 Subscriber Info..................................................................................... - 76 -

3.2.2 Call Monitor........................................................................................... - 77 -

3.2.3 CDR ........................................................................................................ - 79 -

3.2.4 System Info .......................................................................................... - 81 -

3.3 Network ...................................................................................................... - 82 -

3.3.1 Network ................................................................................................. - 82 -

3.3.2 DHCP Srv. (IPv4) ................................................................................. - 85 -

3.3.3 DDNS Srv. (IPv4)................................................................................. - 86 -

3.4 Management ............................................................................................. - 87 -

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3.4.1 Time Setting ......................................................................................... - 87 -

3.4.2 SMTP Setting ........................................................................................ - 89 -

3.4.3 VM Setting............................................................................................. - 90 -

3.4.4 Security.................................................................................................. - 92 -

3.4.5 Firmware Upload.................................................................................. - 94 -

3.4.6 Music Upload......................................................................................... - 95 -

3.4.7 Import Setting...................................................................................... - 96 -

3.4.8 Export Setting ...................................................................................... - 97 -

3.4.9 Rest To Default ..................................................................................... - 98 -

3.4.10 Reboot System.................................................................................. - 98 -

3.4.11 Power Off System............................................................................. - 98 -

CH4. Application Setting............................................................- 99 -

4.1 Customize System prompt................................................................... - 99 -

4.1.1 Record Greeting ................................................................................... - 99 -

4.1.2 Enable Automated Attendant ............................................................ - 99 -

4.1.3 How to record the other System Prompts ...................................... - 99 -

4.2 Customize Ring Back Tone (Transferring Tone) ........................- 114 -

4.3 Call Features............................................................................................- 115 -

4.3.1 Authentication .....................................................................................- 115 -

4.3.2 Automated Attendant.........................................................................- 115 -

4.3.3 Call Transfer.........................................................................................- 115 -

4.3.4 Blind Transfer.......................................................................................- 115 -

4.3.5 Call Forward on Busy .........................................................................- 115 -

4.3.6 Call Forward on No Answer...............................................................- 115 -

4.3.7 Call Forward Unconditional ...............................................................- 115 -

4.3.8 Call Forward Unavailable ...................................................................- 116 -

4.3.9 Call Hold/Retrieval (Client based) ...................................................- 116 -

4.3.10 Call Routing ......................................................................................- 116 -

4.3.11 Call Waiting (Client based)............................................................- 116 -

4.3.12 Caller ID............................................................................................- 116 -

4.3.13 CLIR (Caller Line Identification Restriction)...............................- 116 -

4.3.14 Do Not Disturb (Client based) ......................................................- 117 -

4.3.15 Flexible Extension Logic.................................................................- 117 -

4.3.16 Music On Hold ..................................................................................- 117 -

4.3.17 Music On Transfer............................................................................- 117 -

4.3.18 Call Pickup ........................................................................................- 117 -

4.3.19 Call Park............................................................................................- 117 -

4.3.20 Camp-On (Call Back on Busy) ......................................................- 117 -

4.3.21 Meetme Conference........................................................................- 118 -

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4.3.22 Broadcast..........................................................................................- 118 -

4.3.23 Time and Date .................................................................................- 118 -

4.3.24 Trunk (WG2680)..............................................................................- 118 -

4.3.25 VoIP Gateways (WG2680; WG2504)...........................................- 118 -

4.3.26 Voice Mail to e-mail ........................................................................- 118 -

4.3.27 Access Voice Mail by phone set....................................................- 118 -

4.3.28 Call Monitor ......................................................................................- 118 -

CH5. Appendix.............................................................................. - 119 -

5.1 Voice Mail System Concept.................................................................- 119 -

5.2 System Prompts (Chinese)................................................................ - 122 -

- 4 -

CH1. Overview

SIPPBX 6200A IP-PBX is an IP based IP-PBX which including legend digital PABX

telephony services, Auto Attendant, Voice Mail, Music Ring Back Tone, Conference and

Announcement features together. It also works on both IPv6 and IPv4 network address

simultaneously, which makes migration from IPv4 to IPv6 network smoothly.

SIPPBX 6200A is not only an IP-PBX, but also an efficient communication tool to help your

business and management more efficient. With flexible and full functionality, Welltech

SIPPBX 6200A can give a complete transition from traditional PABX to the new generation

IP-PBX.

1.1 Specifications

¾ Protocol

 SIP RFC 3261 Compliance/ Asterisk Compatible  Support IPv4/IPv6 Dual IP Network Stack

 Network (IPv4): Support Fixed IP, DHCP, And PPPoE mode

 Network (IPv6): Support Fixed And Autoconfig

¾ IVR

 Web Based Auto Attendant Call Rule

 Scheduled And Fixed Greeting  Support Branch Office

 System Prompt Recording By Phone Set

¾ Voice Mail

 Voice Mail To e-Mail

 Voice Mail System

 Message Waiting Indication  Personal Greeting

¾ Toll Restriction

 Provided Different Level Toll Restriction Service

 Support Call Duration Restriction

 Support Personal Password  Support Outgoing Call Routing Password

¾ Call Features

 Flexible Extension Logic

(IPV4 Only)

 CPE Based Call Transfer (Consultant, Blind)

 Server Based Call Transfer (Consultant, Blind)

 Call Forward (Busy, No Answer, Unconditional, Unavailable)

 Call Hold/Retrieval

 Call Routing

 Call Waiting

- 5 -

 Call Pickup (Global, Specific)

 Call Park

 Call Camp-On (Call Back on Busy)

 CLIR (Caller Line Identification Restriction)

 DND (Do Not Disturb)

 Dial Group (Ring All, Sequential Ring, Dynamic Ring)

 Speed Dial

 Music Ring Back Tone

 Music On Hold

 Music On Transfer

 Built-in CDR Report

 Call Monitor

 Broadcasting Service

 Meetme Conference

(6 rooms, 16 members per group)

(6 rooms, 8 members per room)

 Busy Lamp Field (RFC 4235)

¾ Codecs

 G.711 (A-Law & μ-Law)

 G.729

 G.723 Pass-Thru

 GSM

 H.263 Pass-Thru

 MPEG4 Pass-Thru

¾ Technical Features

 Support Subscriber Registration for IPv4 And IPv6

 Support Call Routing Between IPv4 And IPv6

 Support RTP Proxy Between IPv4 And IPv6

 Support T.38 FAX

 Support DDNS

(IPV4 Only)

 Built-in NTP Client/Server

 Built-in DHCP Server

(IPV4 Only)

 Built-in Simple Firewall

 Subscriber NAT Traversal

 Behind NAT Support

 Voice Codec Transform

(IPV4 Only)

(G.711/ G.729/ GSM)

 Auto Provision (Welltech Proprietary) with LP600N IP Phone

 Multiple Language Support ( English and Chinese )

 Management: Web Browser Management

 HTTP Firmware Upgrade

 Export/Import Configuration

 Network Interface: 1WAN, 1LAN

- 6 -

 DTMF: in-band, RFC2833, SIP-Info

¾ Capacity

 200 Concurrent Registers

 50 Concurrent Calls

¾ Dimension



19 Inch Rack Mount

- 7 -

1.2 Hardware Overview

1.2.1 The Front Panel

The SIPPBX 6200A LEDs, which inform you about network activities, are located on the

front panel.

3 2 1

4 5 6

Functions:

¾ 1: System Status LED (not used) ¾ 2: H/D LED ¾ 3: Power LED ¾ 4: Network Interface LED (not used) ¾ 5: Network Interface LED (not used) ¾ 6: Power Switch

1.2.2 The Back Panel

The SIPPBX 6200A ports are located on the back panel.

Functions:

¾ 1: AC Power Outlet ¾ 2: Mouse (not used) ¾ 3: Keyboard

3 4

61 7 9

- 8 -

¾ 4: RS232 Console Port ¾ 5: DVI (not used) ¾ 6: VGA ¾ 7: USB (not used) ¾ 8: Sound (not used) ¾ 9: WAN Interface ¾ 10: LAN Interface

- 9 -

CH2. Start to configure SIPPBX 6200A

2.1 Connection SIPPBX 6200A

Step 1: Connect LAN port of SIPPBX 6200A with PC via crossover cable or connect with

Switch/ Hub via straight through cable.

Step 2: Prepare one computer, and change the IP address to be 192.168.123.12x with

subnet mask 255.255.255.0.

Step 3: Open browser and link to default LAN IP address of SIPPBX 6200A

“192.168.123.123” with default port number 10087, i.e.

http://192.168.123.123:10087

Step 4: Login SIPPBX 6200A with default user ID/Password: “root/root”. After login

SIPPBX 6200A, user can start to configure basic and essential configurations.

Step 5: To configure basic and essential configurations

To make SIPPBX 6200A work has to set some basic and essential configurations,

those include Network, Extension (FXS and IP Phone devices), and Trunk (FXO

devices).

- 10 -

2.1.1 Network Configuration

To change your Network Setting, click Network, and then click the Network Setting

table. The screen appears as shown.

Figure Network: Network Setting

The following table describes the table in this screen

Table Network: Network Setting

Label Description

WAN

IP Support Select IP mode to provide IPv4 only, IPv6 only or dual

IPv4/IPv6.

Mode Select SIPPBX 6200A WAN port network mode to be Fixed IP,

DHCP or PPPoE.

IP Address Enter the IP Address. If user has set SIPPBX 6200A to be fixed

IP mode.

Subnet Mask Enter the Subnet Mask Address. If user has set SIPPBX 6200A

to be fixed IP mode.

Default Gateway Enter the Default Gateway Address. If user has set SIPPBX

6200A to be fixed IP mode.

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Primary DNS Enter the IP address for Primary DNS. The default is

168.95.1.1.

Secondary DNS Enter the IP address for Secondary DNS. The default is null.

Default Gateway

Backup

Select Enable option, if there are any connection problem

occurred on primary default gateway connections, all the traffic

will be guided and switched to the secondary default gateway

for proper operation. The default is Disable.

Secondary Default

Gateway

Enter the Secondary Default Gateway. If you choose the

Default Gateway Backup to Enable.

Check Point Enter the Check Point IP address If you choose the Default

Gateway Backup to Enable. SIPPBX 6200A use the ping

command to PING this IP address in order to check if there are

any connection problem occurred on primary default gateway

connections.

Current used Default

This field display existing used Default Gateway IP address.

Gateway

PPPoE ID Enter the PPPoE ID. If you choose the Mode to PPPoE.

PPPoE PWD Enter the PPPoE Password If you choose the Mode to PPPoE.

IPV6 Mode Select Manual option. You can enter IP Address, Prefix Length

and Gateway address.

IPV6 IP Address Display the IPV6 IP Address. You can enter the IPV6 IP Address,

if you choose the IPV6 mode to Manual.

IPV6 Prefix Length Display the IPV6 Prefix Length. You can enter the IPV6 IP

Address, if you choose the IPV6 mode to Manual.

IPV6 IP Gateway Display the IPV6 Default Gateway Address. You can enter the

IPV6 Default Gateway Address, if you choose the IPV6 mode to

Manual.

MAC This field shows the MAC address. The Mac address cannot be

modified.

LAN (IPV4)

IP Address Enter the IP Address. The default is 192.168.123.123.

Subnet Mask Enter the Subnet Mask Address. The default is 255.255.255.0.

MAC This field shows the MAC address. The Mac address cannot be

modified.

Network Routing Table (IPV4)

Select Select this check box, then modify or delete it.

Destination This field shows the IP address.

Network This field shows the Subnet Mask address.

Gateway This field shows the Default Gateway address.

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Add Click on the Add button, then display Network Router screen.

Modify IP address can be modified by clicking on the checkbox next to

the IP address and click on the Modify button.

Delete IP address can be deleted by clicking on the checkbox next to

the IP address and click on the Delete button.

Apply Click on the Apply button to save your customized settings and

exit this screen.

Cancel Click on the Cancel button to begin configuration this screen.

- 13 -

2.1.2 Extension Configuration

User has to set Extension account for extension devices to register on SIPPBX 6200A.

To change your Extension, click Configuration, and then click the Extension table. The

screen appears as shown.

Figure Configuration: Extension

The following table describes the table in this screen

Table Configuration: Extension

Label Description

Select Select this check box, then modify or delete it.

Extension Number This field shows the Extension Number.

Comment This field shows the Comment information such as user name.

Type This field shows subscriber Type information such as Auto

Configure.

NAT Traversal This field shows the NAT Traversal enable or disable.

Dial Plan This field shows the Dial Plan information includes routes plans.

Add Click on the Add button, then display a new Extension Setting

screen.

Modify An extension can be modified by clicking on the checkbox next

to the extension and click on the Modify button.

Delete An extension can be deleted by clicking on the checkbox next to

the extension and click on the Delete button.

Quick Add Click on the Add button, then display an new Extension Setting

screen.

Quick Delete Click on the Quick Delete button, then display a batch of

Extension number to be deleted.

- 14 -

Total Count This field shows Total subscriber Counts information.

Number Search Enter the search number, then click enter key. The screen will

display match search data.

Total Page This field shows Total Pages of subscriber information.

Page This field shows Page Number information where you are. You

can Enter page number, then click enter key. The screen will

display this page data.

Next/Prev Click on the Next/Prev to Next/Previous Page. The system will

auto display the Next or Previous page Information.

Click Add/Modify. The screen appears as shown.

Figure Configuration: Extension Setting

The following table describes the table in this screen

Table Configuration: Extension Setting

Label Description

Extension Number Assign the number of Extension. This number is also the

Subscriber Type Choose one option to Subscriber Type. Provide drop-down

options: Normal or Autoconfig.

Normal: You must enter Password to this subscriber.

Autoconfig: You must enter MAC Address of this subscriber.

Password Select Normal option to Subscriber Type. Assign the register

- 15 -

password for device to register on SIPPBX 6200A.

Subscriber MAC Select Autoconfig option to Subscriber Type. You must enter

the CPE Device MAC Address such as LP600N IP Phone MAC

address.

Call Group You can use the Call Group parameter to assign an Extension to

one or more groups.

Pickup Group You can use the Pickup Group option in conjunction with this

parameter to allow a ringing phone to be answered from

another extension.

Note:

The Pickup Group option is used to control which Call Groups a

channel may pick up—a channel is given authority to answer

another ringing channel if it is assigned to the same Pickup

Group as the ringing channel’s Call Group. By default, remote

ringing extensions can be answered with *8.

You can define multiple Call Groups and Pickup Groups for one

Extension by a “comma”. For example, you can input “1,3,5”

into Call Group or Pickup Group.

Dial Plan Define the dialing plan for Extension. It specifies the location of

the instruction used to control what extension is allowed to do,

and what to do with incoming calls for this extension. In this

field, you can Choose 5 dial levels for Extension, including

[ext-only], [ext+R1], [ext+R12], [ext+R123], [ext+allroutes].

You can define an “Outgoing call” record, to a certain Route

Level, as R1, R2…, etc. [ext-only] means this subscriber can

only call to Extension. [ext+R1] means the subscriber with

such Dial Plan can call to Extension and Route Level with R1.

[ext+R12] means the subscriber with such Dial Plan can call to

Extension and Route Level with R1 and R2. [ext+R123] means

the subscriber with such DialPlan can call to Extension and

Route Level with R1, R2 and R3. [ext+allroutes] means the

subscriber with such Dial Plan can call to Extension and Route

Level with R1, R2, R3 and R4(allroutes).

Note:

For more information about Route Level, please refer to the

user manual:

CH3.1.8 Outgoing Routing.

Keypad User can select Keypad type to be RFC2833, In-band, SIP-Info

and Auto. You can choose Auto to auto select the Keypad type.

Choose RFC2833, Inband or SIP-Info here will force the

Extension use RFC2833, Inband or SIP-Info accordingly and

- 16 -

the setting should be also matched the Keypad setting of

Extension device.

Note:

Now SIPPBX 6200A could not support G729 with Inband

Keypad type. If SIPPBX 6200A detect the calling party or

receiving party do not support RFC2833 DTMF type, SIPPBX

6200A will switch the voice Codec to G.711 to make sure the

DTMF detection is correctly.

NAT Traversal If the Extension device is installed behind a device performing

NAT such as firewall or router, and need to register to SIPPBX

6200A on public network, this extension has to enable this

function. Enable NAT Traversal to force SIPPBX 6200A to ignore

the contact information for the Extension and use the address

from which the packets are been received.

Fixed Trunk ID User can define a Fixed Trunk for a certain extensions. When

such extension makes an outgoing call via routing table,

SIPPBX 6200A will check “Fixed Outgoing Call Rule” first. If

“Fixed Outgoing Call Rule” is enabled, SIPPBX 6200A will

confirm the Fix Trunk ID for the calling party. That means the

outbound call will be routed by Fixed Trunk ID, if you define the

Fixed Trunk ID for the calling party and you also enable “Fixed

Outgoing Call Rule”.

Note:

For more information about Fixed Outgoing Call Rule, please

refer to the user manual:

CH3.1.8 Outgoing Routing.

Absolute Timeout Specific the timeout value for the outgoing calls. Please also go

to Outgoing Call Rule page to enable the Route Timeout

function to restrict talking time ( Toll Restriction ).

BLF Enable Busy Line Field function for extensions.

Forward CallerID By default, the “from header of SIP invite” will contain the

caller’s line number when forward function is activated. But this

may make some errors occurred for some SIP Trunk services.

So we add this function in the “Extension Setting” page, to let

user modify the line number of SIP Invite’ s from header, from

calling party’s number to the called party’s number.

Unconditional FWD Enable Unconditional forward function by adding forwarding

number.

No Answer FWD Enable No Answer forward function by adding forwarding

number.

- 17 -

Busy FWD Enable Busy forward function by adding forwarding number.

Unavailable FWD Enable Unavailable forward function by adding forwarding

number.

Comment You can input a 20 bytes note for each extension here.

Mail Box User can select to disable or enable voice mail box function. If

this function is enabled, user could input e-mail address for the

Extension. When having voice mail of incoming call left, system

will send this voice mail to the specified e-mail address. You can

also login the mail box system by dialing to *98.

E-Mail Address This field will appear when you enable Mail Box function and

you can input the E-Mail Address here for voice mail to E-mail.

Note:

Please remember to set the SMTP in the page of Management

Web, and then click SMTP Setting to activate the Voice Mail to

E-mail.

If the SIPPBX 6200A got a new message left at one subscriber,

it sends the message to the user by email immediately. If you

are using SIPPBX 6200A and you want the SIPPBX 6200A to

save voice mail to it and not send to email. You just need to

input “x” to E-Mail Address.

Save VM to Local If you select Enable to Save VM To Local when you have Voice

Mail message, this message will be saved at Local fresh

memory folder.

VM Login Password SIPPBX 6200A has a built-in voice mail system. And user can

mailbox number and password to retrieve voice mail. User can

define the Voice Mail box login password here. Another way to

For example, dial *98101 can login extension 101 voice mail

box, and caller can enter password to access voice mail.

Voice Mail Count Display the exact count of New Messages and Old Messages.

Delete MailBox

Content

User can delete all of the voice mails and personal greeting by

marking the “Delete MailBox Content” and then press Apply.

Apply Click on the Apply button to save your customized settings and

exit this screen.

Cancel Click on the Cancel button to begin configuration of this screen.

- 18 -

2.1.3 Trunk Configuration

User has to set Trunk account for Trunk (FXO device, for instance, WellGate 2540 or 2680 )

to register to SIPPBX 6200A or set some necessary configuration for SIP trunk.

To change your Trunk, click Configuration, and then click the Trunk table. The screen

appears as follows.

Figure Configuration: Trunk

The following table describes the table in this screen

Table Configuration: Trunk

Label Description

Select Select this check box and modify or delete it.

Trunk Number This field shows the Trunk Number information.

Comment This field shows the remark information.

NAT Traversal This field shows the NAT Traversal information.

Maximum Channels This field shows the Maximum concurrent call to this trunk.

There is no limitation if this field left blank.

Add Click on the Add button to display a new Trunk Setting screen.

Modify A Trunk can be modified by clicking on the checkbox next to the

Trunk and click on the Modify button to modify its contents.

Delete A Trunk record can be deleted by clicking on the checkbox next

to that Trunk and click on the Delete button.

Total Count This field shows Total Counts of trunk information.

Number Search Enter the search number, then click enter key. The screen will

display matched search data.

Total Page This field shows Total Pages of trunk information.

- 19 -

Page This field shows Page Number information. You can Enter page

number, then click enter key. The screen will display this page

data.

Next/Prev Click on the Next/Prev to Next/Previous Page. The system will

auto display the Next or Previous trunk Information.

Example 1: Set Trunk for FXO gateway

Click Add/Modify. The screen appears as follows.

Figure Configuration: Trunk Setting

The following table describes the table in this screen

Table Configuration: Trunk Setting

Label Description

Trunk Number Assign the number of Trunk. This number is also the register

name for Trunk device.

Note: The Trunk Number can also be a “Trunk ID”. In the

Routing Table page, you should define the destination of prefix

route. When you define the prefix route, you should set the

Trunk ID (Trunk Number) in the Trunk page first; then you

could input the correct Trunk ID in the Destination field.

Password Assign the register password for device to register on SIPPBX

6200A.

Host Setting the Host to Dynamic will require the trunk to register

the SIPPBX 6200A so that the SIPPBX 6200A know how to

reach the trunk. You can also set the Host to an IP address or

FQDN (domain name) if you set the Host to [Pre-define]. There

will be a field called [Address] appeared when you choose Host

to [Pre-define]. This limits only where you place calls to, as the

user is allowed to place calls from anywhere.

- 20 -

Dial Plan Define the dialing plan for Trunk. It specifies the location of the

instruction used to control what the phone is allowed to do, and

what to do with incoming calls for this Trunk. In this field, you

can Choose 6 dial levels for Extension, including [from-pstn],

[ext-only], [ext+R1], [ext+R12], [ext+R123], [ext+allroutes].

You can define an “Outgoing call” record to a certain route

level, such as R1, R2…, etc. [from-pstn] is used for Trunk only.

[ext-only] means this subscriber can only call to Extension.

[ext+R1] means the subscriber with such Dial Plan can call to

Extension and Route Level with R1. [ext+R12] means the

subscriber with such Dial Plan can call to Extension and Route

Level with R1 and R2. [ext+R123] means the subscriber with

such Dial Plan can call to Extension and Route Level with R1, R2

and R3. [ext+allroutes] means the subscriber with such Dial

Plan can call to Extension and Route Level with R1, R2, R3 and

R4 (allroutes).

Note: For more information about Route Level, please refer to

the user manual: CH3.1.8 Outgoing Routing.

Keypad User can select Keypad type to be RFC2833, In-band, or

SIP-Info and Auto. You can choose Auto to auto select the

Keypad type. Choose RFC2833, Inband or SIP-Info here will

force the Extension to use RFC2833, Inband or SIP-Info

accordingly and the setting should be also matched the Keypad

setting of Trunk device.

NAT Traversal If the Trunk device is behind a device performing NAT, such as

firewall or router, and need to register to SIPPBX 6200A on

public network, then user has to enable this function. Enable

NAT Traversal to force SIPPBX 6200A to ignore the contact

information for the Trunk and use the address from which the

packets are been received.

Port You can use this to define the SIP signal port if you want to

listen on a nonstandard SIP signal port.

External Server

Address

This field will allow you to set the domain in the SIP From URI.

Setting this will avoid some unexpected issue if the service

provider needs this for authentication.

Maximum Channels This will limit the maximum channels for this Trunk. For

example, you set 2 into this field; only 2 outgoing calls could go

via this Trunk. Default is no limit.

Outbound Caller ID Some service provider will require the correct registered caller

ID if it got an incoming call. SIPPBX 6200A will send the

- 21 -

Extension’s caller ID to this Trunk as default value, if you set

empty here.

Note:

z Normally, SIP From URI contains the Extension’s calling ID

and SIPPBX 6200A’s IP address, but some ITSP may reject

this call due to some security issue. You can modify the

Calling ID and IP/ Domain in the fields of [External Server

Address] and [Outbound Caller ID] when the call is going

via the SIPPBX 6200A to the Destination (Trunk) to avoid

such security issue.

z If you set a Welltech FXO gateway as the Trunk, you can

use the default Trunk 888 and 889 as the FXO’s register

number.

z For the FXO gateway, you may just configure Trunk

Number, Password, Host, Dial Plan, Keypad, NAT Traversal

and RTP Mode.

z If you set the ITSP as the Trunk, you may need to set the

following configure: Port, External Server Address and

Outbound Caller ID.

Caller ID Display

Name

Comment You can input a 10 bytes note for each Trunk here.

CLIR Support CLIR means "Caller Line Identification Restriction". It is a

Apply Click on the Apply button to save your customized settings and

Cancel Click on the Cancel button to begin a new configuration on this

When inbound call is coming from Trunk, such as 888. The

caller ID Name will be the “PSTN number” or “888”. Specify this

will use the current setting instead.

proper noun. It is a feature to hide the caller's number. For

example, ext 101 call to ext 102. But 101 won’t like to show the

caller ID to 102. So 101 can activate this feature to hide the

caller ID. When 102 got a call from 101, the LCD of 102 should

display "Anonymous".

exit this screen.

screen.

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CH3. Full Web Configurations

After Login SIPPBX 6200A, you will see screen as below, and there are four main

categories, user can click on each category to extend detail items.

 Configuration: Include all telephony configurations of SIPPBX 6200A.

• IP PBX

• Office Call Rule

• Feature Code

• Extension

• Auto Config (IPv4)

• Trunk

• SIP Trunk Reg.

• Outgoing Routing

• Incoming Routing

• Dial Group

• Speed Dial

• Broadcast

• Meetme Conf.

• T.38 FAX

 Information: To show related information.

• Subscriber Info.

• Call Monitor

• CDR

• System Info.

 Network: To show related information.

• Network

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• DHCP Server. (IPv4)

• DDNS Server. (IPv4)

 Management: Include all system management of SIPPBX 6200A.

• Time Setting

• SMTP Setting

• VM Setting

• Security

• Firmware Upload

• Music Upload

• Import Setting

• Export Setting

• Reset To Default

• Reboot System: To reboot system of SIPPBX 6200A.

• Power Off System

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3.1 Configuration

The Network screens can help you configure IP PBX, Office Call Rule, Feature Code,

Extension, Auto configure (IPV4), Trunk, SIP Trunk Reg., Outgoing Routing, Dial Group,

Speed Dial, Broadcast, Meetme Conf., T.38 FAX.

3.1.1 IP PBX

To change your IP PBX Setting, click Configuration, and then click the IP PBX table. The

screen appears as follows.

Figure Configuration: IP PBX

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The following table describes the table in this screen

Table Configuration: IP PBX

Label Description

SIP Setting

IP-PBX Realm Enter the IP-PBX Realm of SIPPBX 6200A. This parameter is

essential when there is more than one SIPPBX 6200A, and user

wants to have inter-calls between SIPPBX 6200A. Please refer to

SIP Trunk configuration.

IP-PBX User Agent Enter the IP-PBX User Agent. IP-PBX User Agent takes as its

argument a string specifying the value for the user agent field

in the SIP header. The default value is IP-PBX.

Proxy Port Enter the Proxy Port. These optional parameters allow you to

control the port on which you wish the SIPPBX 6200A to accept

SIP connections. Default is 5060.

RTP Port Start Enter the RTP Port Start. The voice media will use RTP as the

transport protocol. You can define the RTP port range that

SIPPBX 6200A opened. Default start port is 10000.

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RTP Port End Enter the RTP Port End. The voice media will use RTP as the

transport protocol. You can define the RTP port range that

SIPPBX 6200A opened. Default end port is 20000.

Note: Default RTP port range is 10000 to 20000 and default

proxy port is 5060. If your SIPPBX 6200A is behind a firewall,

please make sure you have already opened the RTP port

(10000-20000) and proxy port (5060). And you should also

make sure the proxy port (5060) has already mapped to

SIPPBX 6200A.

Max Expire Time This sets the maximum amount of time, in seconds. This is

used for the registration expiry time. If this value is less than

the expiry time from the client, and then click the SIPPBX

6200A will reply a certain expiry time which is defined in

“Default Expire Time” to client.

Default Expire Time This sets the default SIP registration expiry time, in seconds. A

client will normally define this value when it initially registers,

so the default value you set here will be used only if the client

does not specify a timeout when it registers. If you are

registering to another SIP Trunk, this is the registration timeout

that it will send to the far end.

Codec Priority

Codec Priority Codec negotiation is attempted in the order in which the voice

Codec Priority is defined. Default is G.729 with the first priority,

G.711u with second priority, G.711A with third priority and

GSM is fourth priority. That means the SIPPBX 6200A can only

recognize these four Codecs and it will force the voice Codec

with the specified priority and forward to another subscriber.

Now, SIPPBX 6200A can support G.729, G.711U, G.711A, GSM

and G.723 Pass-Thru.

PBX Setting

Ext Ring Time This field defines the timeout value if the call is made between

Extension to Extension. Default is 20 seconds.

Out Ring Time This field defines the timeout value if the call is made from

Extension to outside Line (defined by routing table). Default is

no limitation timeout.

Music RBT If this call was made between extensions. Enabling this option

will provide music to the calling party as Ring Back Tone until

the call was answered.

Music RBT (After AA) If this call was forwarded from Auto Attendant. Enabling this

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option will provide music to the calling party as Ring Back tone

until the call was answered.

Call Monitor Refresh SIPPBX 6200A has call monitor function. The call situation will

be refreshed by the refresh times. Default is 30 seconds and

user can change it here.

RTP Timeout It terminates a call if no RTP data received within the time

specified.

Video Support This field will enable video call with H263 pass-through or

MPEG4 pass-through.

Video Format Choose one option to Video Format. Provide drop-down

options: H263 pass-through or MPEG4 pass-through.

SRVlookup Support Enable or disable SRV lookup. DNS SRV is a way of setting up a

logical, resolvable address where you can be reached. This

allows calls to be forwarded to different locations without the

need to change the logical address, but your DNS Server must

support it as well. If you are not sure, please disable it. This

option is Disable, Enable by default.

Invalid Number

Support

Normally, a busy tone will be heard if caller dial to a non-exist

number. Enable this option, the caller should hear an

announcement to notify that this number is not existed.

Behind NAT

Behind NAT If your SIPPBX 6200A is installed behind NAT, we strongly

suggest you to enable Behind NAT to avoid some unexpected

issues, such as “one way voice”.

External IP If you enter External IP address, SIPPBX 6200A will take that IP

address as its argument. If SIPPBX 6200A is behind NAT, the

SIP header will normally use the private IP address assigned to

the server. The remotely device does not know how to route

back to this address; therefore, it must be replaced with a

valid, routable address.

Secondary External

This should cooperate with Default Gateway Backup in Network

page. When Default Gateway Backup is enabled, the SIPPBX

6200A will auto switch the default gateway to secondary one if

primary default gateway is broken. The External IP is not

functional when Backup Default Gateway is chosen. So you

must enter the Secondary External IP for Backup Default

Gateway.

External Host External Host takes a fully qualified domain name as its

argument. If SIPPBX 6200A was installed behind NAT, the SIP

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header will normally use the private IP address assigned to the

server. If you set this option, SIPPBX 6200A will perform

periodically DNS lookups on the hostname and replace the

private IP address with the IP address returned from the DNS

lookup.

Note: You should not set both of External IP and External Host

together; otherwise there will be some unexpected problems

appeared. That means you can only choose either External IP

or External Host for “Behind NAT”.

Local Net Local Net is used to tell SIPPBX 6200A which IP addresses are

considered local. If one of caller or callee is not under Local Net,

SIPPBX 6200A will set the address in the SIP header that can be

translated to that specified by External IP or the IP address can

be looked up with External Host. The format will be IP/ Subnet

Mask. Example: 192.168.1.0/ 255.255.255.0

Apply Click on the Apply button to save your customized settings and

exit this screen.

Cancel Click on the Cancel button to begin configuration this screen

afresh.

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3.1.2 Office Call Rule

You can define a business time to forward incoming call to company announcement or a

certain destination.

6200A provide 12 kinds of office call rule (**998 to **988). User can setup a FXO gateway

and hotline to **999 (for office 1) or **988 (for office 12) to reach auto attendant.

Now user can make SIPPBX 6200A to decide the destination when it got an invite with

called number as **999 to **988. When SIPPBX 6200A got an invite with **999, SIPPBX

6200A will confirm the current time and forward this call to AA, Ext, Group or Outbound. If

you choose the destination to EXT, Group or Outbound, please remember to input the

destination number into the following field.

When you set the destination to AA, please refer to CH4.1.3 How to record the other

system prompts for the greeting recording.

To change your Office Call Rule, click Configuration, and then click the Office Call Rule

table. The screen appears as shown below.

Figure Configuration: Office Call Rule

The following table describes the table in this screen

Table Configuration: Office Call Rule

Label Description

Select Select this check box, then modify it.

Representative No. This field shows the Representative number information

Comment This field shows the Comment information.

Operator This field shows the Operator information.

Call Rule This field shows the Call Rule information.

Modify A Representative number can be modified by clicking on the

checkbox next to the Representative number and click on the

Modify button.

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+ 102 hidden pages

Welltech SIPPBX-6200A User Manual

Specifications and Main Features

Frequently Asked Questions

User Manual

1.1 Specifications

1.2 Hardware Overview

1.2.1 The Front Panel

1.2.2 The Back Panel

CH2. Start to configure SIPPBX 6200A

2.1 Connection SIPPBX 6200A

2.1.1 Network Configuration

2.1.2 Extension Configuration

2.1.3 Trunk Configuration

CH3. Full Web Configurations

3.1 Configuration

3.1.1 IP PBX

3.1.2 Office Call Rule