Copyright (C) 2008 PLANET Technology Corp. All rights reserved.
The products and programs described in this User’s Manual are licensed products of PLANET
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Disclaimer
PLANET Technology does not warrant that the hardware will work properly in all environments and
applications, and makes no warranty and representation, either implied or expressed, with respect to
the quality, performance, merchantability, or fitness for a particular purpose.
PLANET has made every effort to ensure that this User’s Manual is accurate; PLANET disclaims
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CE Declaration of conformity
This equipment complies with the requirements relating to electromagnetic compatibility, EN 55022
class A for ITE and EN 50082-1. This meets the essential protection requirements of the European
Council Directive 89/336/EEC on the approximation of the laws of the Member States relating to
electromagnetic compatibility.
The is a class B device, In a domestic environment, this product may cause radi o interference, in which
case the user may be required to take adequate measures.
ii
FCC Notice
This equipment has been tested and found to comply with the limits for a Class A digital device,
pursuant to Part 15 of FCC Rules. These limits are designed to provide reasonable protection against
harmful interference when the equipment is operated in a commercial environment. This equipment
generates, uses, and can radiate radio frequency energy and, if not installed in accordance with the
instruction manual, may cause harmful interference to radio communication. Operation of this
equipment in a residential area is likely to cause harmful interference in which case the user will be
required to correct the interference at the user’s own expense.
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presence of hazardous substances in electri cal and el ectronic equipment, end users of
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The PLANET logo is a trademark of PLANET Technology. This documentation may refer to numerous
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Revision
PLAENT IP PBX User’s Manual
Revision: 1.0 (June. 2008)
9.3 IP PBX Voicemail System Menu Tree.......................................................................106
xi
1 Introduction
1.1 Overview
PLANET IPX-2000 series IP PBX system are designed and optimized for the enterprise, and SMB
daily communications. The IPX-2000 is the next generation voice communication plat form for the small
to medium enterprise. Designed as an open, scalable, and highly reliable telephony solution, the
IPX-2000 series are able to accept 200 extension registrations, and effectively meeting scales from
various enterprises. Designed to run on a variety of VoIP applications, the IPX-1800 and IPX-2000
provides centralized call control, auto-attendant, voice conferencing, PSTN, and IP-based
communications.
The IPX-2000 integrates up to 8 FXO to become a feature-ric h PBX system that supports seamless
communications between existing PSTN call s, analog, IP phones and SIP-based endpoints
The IPX-2000 system integrates telephony call processing , call control, voice mail, and a widel y PBX
application programming interface into a hig hly scalable architecture designed to s upport both
traditional circuit-based and the Internet telephon y service within a distributed enterprise
communications network.
With IPX-2000 system, s tandard SIP phones can be easily integrated i n your office; plus the
auto-config feature, you may integrate our IP Phone series - VIP-254T/VIP-254T, and the ATA (analog
telephone adapter) series - VIP-156/VIP-157 to build up th
Allowing distributed IP technology to meet traditional voice services, with proactive management
interface, the IPX-2000 system in the daily business processes, enterprises can make people more
productive, more intelligent tasks, and more customer satisfaction.
e VoIP network deployment in minutes.
1.2 Physical Interfaces
Figure 7. Front Panel of IPX-2000
Figure 8. Rear Panel of IPX-2000
Power cord
Telephony interface ports
USB ports
WAN
LAN
Table 1. Physical interfaces description of IP PBX
100-240 Volt, 50-60 Hz
IPX-2000 2 x slots integrates up 8 FXO
4 ports per daughter card, numbered from right to
the left. The rightmost port at slot 0 is port 1 and
FXO module
the leftmost port at slot 1 (if installed) is port 8. FXO
ports are to be connected to FXS jacks on wall or
analog PBX using RJ-11 cables.
1 external port with compliance to USB 1.1/2.0. Plug in a USB
hard drive for voicemail backup from the internal one
Connect to a broadband modem or a WAN router
Connect to a LAN switch
2
2 Managing with Web Interface
The factory default of LAN IP address is 192.168.1.1. Connect to LAN port and the configuration We b
interface is at https://192.168.1.1/
Click Yes to see the home page. Ty pe in the default administ rator I D and p assword (both are admin) to
log in for administration. The administrator password can be changed in the User Management ->
User.
1. Click admin in the Login ID.
2. Change the password in Password.
3. Click Apply to change the password.
Note: For the system security, please change the password after the first log-in.
After login, you will see four icons, Add Account Wizard, Add Route & Trunk, Mass Extensions Adding and Customize Setup. The first three icons can lead you step by step to configure some ba sic
settings of IP PBX. Click the Customize Setup icon to see all the PBX configurations into detail.
Administrator can click
Web Interface.
. Once connected, the browser will ask for accepting a certificate.
on the top-left side of the webpage to go back to the home page of IP PBX
3
3 Wizard Configuration
With IP PBX Wizard configuration, the administrator can set basic configurations for IP PBX easily.
With basic setup, IP PBX can function, and connect to the relevant devices and trunks. The Wizard
Configuration including Add Account Wizard, Add Route & T runk and Mass Extension Adding.
When entering Wizard configuration, you will see
you to configure with Wizard. Any configuration change in Wizard requires clicking
bottom of the homepage.
at the bottom of each page that helps
at the
3.1 Add Account Wizard
In Add User & Device page, the administrator can setup usergro ups, users and devices. You can
follow the following steps to finish configuration. After finishing configuration, click
bottom of the homepage to take the configuration effect.
Step 1: Add User Group
1. Enter a group ID and then click ADD.
2. The name will show in the table of the webpage.
3. Click the name to view the edit page.
at the
4. Enter settings shown in Table 3-1.
5. Click Back to return to the ADD USER GROUP page.
For deleting a usergroup, select a group ID and click DEL. Note: Make sure there is no user associate with the usergroup, or it cannot be deleted.
Click Next to add a user .
Table 3-1Add Usergroup Settings
Field Description
Description
Arbitrary description information. Click SET to add/update
4
Associated Trunk s
the information.
1
Select routegroups and outbound trunks accessible by this
usergroup. Note the order matters the hunting sequence in
run-time.
Routegroup: display available routegroups.
Trunk: Display available trunks.
Group ID: The default number is “0”. A trunk with Group ID
“0” does not form a balance group with any other trunks in
Group 0. If Group ID is 1~9, trunks with the same Group ID
form a usage balance group.
Weight: the weight of a trunk to be selected in a trunk
balance group for an outgoing call.
Reachable User Groups
Associated PBX Features
Click
or to add or delete the associate trunks.
After add all trunks, click APPLY.
) If there is not any appropriate SIP trunk and PSTN
trunks to select, you may assign trunks at Setp 5:
Assign Trunk in Add Route & Trunk wizard
configuration after trunks are created in the previous
step.
Select a usergroup and click ADD that is reachable from this
usergroup. By default, only users in the same usergroup can
reach one another .
) If there is not any appropriate usergroup to select,
come back later to revise this selection, once more
usergroups have been created.
2
Select PBX features enabled to this usergroup.
Member List Show the users associated with this usergroup.
Auth. Dial Passcode Select and enter a password in number for caller to have the
same privilege as this usergroup to dial out.
1
Please refer to 6.6, 6.7, and 6.8 for detail s.
2
Please refer to 7 for details.
5
Setp 2: Add User
1. Enter settings shown in Table 3-2.
2. Click ADD to see the user information in the table of the webpage.
For deleting a user, select a Login ID and click DEL.
Click Next to choose a device.
Table 3-2 Add User Settings
Field Description
Login ID A unique ID containing alphabets, numbers, and underscore
only without spaces; 20 characters maximum. This is the ID
for personal configuration through IP PBX Web
management.
Name Name of the user , either a real or a virtu al one, e.g. Alice Lee
or Conference Room.
Password Password for the user to access IP PBX Web management.
Description Arbitrary description information.
E-mail Address E-mail address of the user for voicemail notification.
Attach Voicemail in E-mail Notification Select to enclose the message received in the notification
e-mail as an attachment.
Usergroup Select the usergroup this user belongs to.
) If there is not any appropriate usergroup to select,
come back later to revise this selection if no
appropriate usergroup could be chosen for now.
Step 3: Choose Device
Based on the devices you have, click ADD IP PHONE or A DD ANALOG PHONE, and Next to add/set
the device.
Note: If selecting ADD ANALOG PHONE, the wizard will skip to Step 5.
6
Step 4: Add Device
1. Enter a device name in the Device ID box.
2. Select Auto Provision if you want to enable Automatic Client Configuration.
3. Click ADD to see the newly added device in the table of the webpage, or to see the Enable
Automatic Client Configuration (ACC) page if Auto Provision is selected.
Enter settings shown in Table 3-3 ACC (Automatic Client Configuration) Settings and click ENABLE. Note: Consult with your vendor to make sure your SIP phone support Auto Provision function.
For deleting a device, select a device ID and click DEL. Note: Make sure there is no extension associate with the device, or it cannot be deleted.
Click Next to set a device.
Vendor Prefix Ask your IP Phone vendor for the Prefix. e.g. eip7012.
MAC Address MAC address of the device.
Supplementary Configuration
Codec Preference
Enable Voice Activity Detection
(VAD)
Supplementary configuration files for IP Phone. The file name
must start with “psc-“.
Preference order of supported codec and packet times of the
phone.
VAD is a technique that detects absence of audio and con se rves
bandwidth by preventing the transmission of "silent packets" over
the network.
) Select if your IP Phone supports VAD.
DTMF mode Choose a DTMF mode used by the phone.
Setp 5: Add Extension
3.1.5.1 Add Extension of IP Phone
1. Enter settings shown in Table 3-4.
2. Click ADD to see the newly added extension in the table of the webpage.
7
For deleting an extension, select an extension number and click DEL.
Click Finish to finalize all the settings, and go back to the homepage.
Table 3-4 Add Extension of IP Phone Settings
Field Description
Extension Number A unique line number composed of digits only, e.g. 101; 20
digits maximum. This is the login ID on the device
configuration side.
Associated Device Select the Device this extension asso ciates with.
3
User
Select the user this extension associates with.
) If there is not any appropriate users to select, one can
come back later once the expected user has been
added.
Password Password of this extension. Same password must be
configured on the device side as well.
Pickup Group The usergroup that the extension can pick up. The
extension can set a usergroup that when any extension in
the usergroup rings, the extension can press *8 to pick up
the call in ringing state.
Select Include Reachables check box to be able to pickup
calls that belong to other usergroups which is configured in
Reachable User Groups of the selected usergroup.
Language Preferred language for system instructions heard from the
extension.
Voicemail Select enable to allocate voicemail account for the
extension.
Voicemail PIN PIN to access voicemails. This is mandatory if above
voicemail option is enabled.
Max Voicema il Space Enter maximum space in KBytes for voicemail. Enter 0 or
leave it as blank for not limiting the voicemail space. The
3
Please refer to 6.1 for details.
8
voicemail will be recorded until the storage is full.
Disable Fast Bridging Select to disable express media forwarding.
) With Fast Bridging feature enabled, if the two parties
involved in a call (for example, one IP extension and
one SIP trunk) use different DTMF modes
(RFC2833/SIP INFO/Inband), inline transfer (*#) or
2nd-dialing might fail. To avoid such problem, it is
recommended to set the same DTMF mode for all IP
extensions and SIP trunks in the IPBX, as well as for
all IP phones registered to the IPBX. If it is not
feasible to set the same DTMF mode for some IP
extensions or SIP trunks, and inline transfer or
2nd-dialing is necessary for those IP extensions or
SIP trunks, the Fast Bridging feature can be disabled
on a per IP extension and per SIP trunk basis. Note
that Fast Bridging is enabled by default.
Try Peer-to-peer RTP
DTMF Mode Choose preferre d DTMF mode for this extension. Currently
If click YES, IP PBX will attempt to notify the two peers in a
conversation to try peer-to-peer RTP tra nsmission. This is
suggested as long as phones support INVITE or UPDATE
method during a connected call to save the resource of IP
PBX. However, only SIP INFO DTMF mode phones should
enable this since other DTMF modes require IP PBX being
RTP relay server to support in-line transfer.
supported types include RFC2833, SIP INFO, and in-band
tone. It must match configuration on the device side.
) In-band DTMF mode consumes the limited DSP
resource when using a highly compressed codec,
such as G.729 or G.723.1. Therefore, calls will not
connect with such setting if DSP is not installed.
Although using a low-complexity codec such as G.711
does not require DSP, DTMF detection still takes
considerable CPU resource and impacts several
system specs. Be cautious when configuring an
extension with in-band DTMF mode.
9
3.1.5.2 Add Analog Phone
1. Enter settings shown in Table 3-5.
2. Click ADD to see the newly added analog phone in the table of the webpage.
For deleting an analog phone, select a POTS port and click DEL.
Click Finish to finalize all the settings, and go back to the homepage.
Table 3-5 Add Analog Phone Settings
Field Description
POTS Port FXS port index.
Pickup Group The pickup group that the extension belongs to. Select
Include Reachables check box to be able to pickup calls
that belong to other usergroups which is configured in
Reachable User Groups of the selected usergroup.
Extension Number A unique line number composed of digits only, e.g. 101; 20
digits maximum.
Unavailable T imeout Timeout for ringing before a call is a nswered.
4
User
Select a user that this extension associates with.
) If there is not any appropriate users to select, one can
come back later once the expected user has been
added.
Language Preferred language for system instructions heard from the
extension.
Voicemail
Voicemail PIN PIN to access voicemails. This is mandatory if above
Select Enable to allocate voicemail account for the
extension.
voicemail option is enabled.
Max Voicema il Space Enter maximum space in KBytes for voicemail. Enter 0 or
leave it as blank for not limiting the voicemail space. The
4
Please refer to 6.1 for details.
10
voicemail will be recorded until the storage is full.
3.2 Add Route & Trunk
In Add Route & T r unk page, the administrator can setup routes, routgroups and trunks. Moreover,
worktime and IVR are included in this part for assigning to trunks. You can follow the following steps to
finish configuration. After finishing configuration, click
take the configuration effect.
at the bottom of the homepage to
Step 1: Add Route
1. Enter settings shown in Table 3-6.
2. Click ADD to see the newly added route in the table in the webpage.
For deleting a route, select a route ID and click DEL.
Click Next to set a routegroup.
Table 3-6 Add Route Settings
Field Description
Route ID A unique ID containing alphabets, numbers, and underscore
only without spaces; 16 characters maximum.
Description Arbitrary description information.
Destination Number Pattern
Prefix A sequen ce of digit s to be prefixed to the final dialed number
5
For more information about the available digit set and wildcard characters, please refer to Table 6-7.
5
A destination number pattern consisting of digits, digit set,
and wildcard characters, e.g. 9NXXXXXX matches any
7-digit called number starting from a digit larger or equal to 2
and with an extra prefix digit 9.
after stripping. Using 9NXXXXXX as an example route
pattern with number of stripped digits equal to 1 and prefix
1408, dialing 95270001 will be 14085270001 when it
actually got dialed out.
A special pref ix character “w” could be used for PSTN trunks
to pause 0.5 second during dialing. Say, 4 leading
11
consecutive “w” result in 2 seconds delay before dialing.
Number of Stripped Digits Select number of leading digits to be stripped from the
original dialed number when matches this route. Using
9NXXXXXX as an example route pattern with number of
stripped digits equal to 1, dialing 95270001 will be stripped
to be 5270001 when it actually got dialed out.
Setp 2: Add Route Group
1. Enter a group ID and then click ADD.
2. The name will show in the table of the webpage.
3. Click the name to view the edit page.
4. Enter settings shown in Table 3-7.
5. Click Back to return to the ADD ROUTE GROUP page.
For deleting a routegroup, select a group ID and click DEL.
Click Next to choose a trunk.
Note: Make sure there is no route associate with the routegroup, or it cannot be deleted.
Table 3-7 Add Routegroup Settings
Field Description
Description
Arbitrary description information. Click SET to add/update
the information.
Associated Routes
6
Select routes belonged to this routegroup. Click or
button to add or remove a route to or from the
routegroup. The right box lists current selected routes. Click
SET to update the information. Note the order of the
selected routes is important since it decides which route
would be matched first for an outgoing call.
) If there is no appropriate routes to select initially, one
6
Please refer to 6.4 for details.
can come back later to revise it, once the expected
routes are added.
12
Setp 3: Choose Trunk
In the Choose T runk page, click SIP TRUNK, ANALOG PSTN TRUNK or ISDN PSTN TRUNK to see
one of the following pages to add various types of trunks.
Setp 4: Add Trunk
3.2.4.1 Add SIP Trunks
1. Enter settings shown in Table 3-8.
2. Click ADD to see the newly added SIP trunk in the table in the webp age.
For deleting a SIP trunk, select a trunk identifier and click DEL.
Click Next to assign trunks to usergroups.
Table 3-8 Add SIP Trunk Settings
Field Description
Trunk Identifier A unique num ber consisting of digits only. Usually give the
phone number issued by the ITSP for consistency.
Description Arbitrary description information.
Auth. Name
Auth. Password Give the password used for authentication on the remote
Dynamic Peer Select if the trunk is a passive trunk which means the
Specify the name for authentication if different to the Trunk
Identifier.
SIP proxy or registrar. Usually this is given by the ITSP.
registration will be from a dynamic remote peer. T y pical
application is to accept registration from an IP PBX at a
remote site with dynamic IP address. Once the remote IP
PBX registers, calls from local to remote can be made
reversely over the trunk.
SIP Proxy IP
SIP Proxy Port
Registration Required Select if registration to a registrar is required to activate the
Specify IP address (or fully qualified d omain name) a nd UDP
port of the remote SIP proxy, which usually refer to the SIP
server on the ITSP side.
trunk. This is true for a remote IP PBX or an ITSP account,
13
however, may be not required in case of a SIP gateway.
SIP Registrar IP
SIP Registrar Port
Language Preferred language for system instructions heard from the
DID T ype DID means direct inward dialing (also called DDI in Europe).
None DID
Extension DID
Specify IP address (or fully qualified d omain name) a nd UDP
port of the remote SIP registrar, which usually refer to the
SIP server on the ITSP side (same as proxy).
trunk.
Select a preferred type, None DID, Extension DID, DID by
NumberDID by Privilege or Centrex DID from the list and
then enter configuration in DID Prefix and DID Stripping to
have the incoming calls directed to the corresponding trunk.
When selected None DID, all incoming calls will enter IVR
system instead of directly dial to a specified extension.
Select a preferred IVR from IVR list for this trunk.
When selected Extension DID, select an extension in the
list to be an unconditional destination for incoming calls to
this trunk. If prefix or stripping has been given, the result of
DID By Number
DID By Privilege
digit manipulation is dialed in a DTMF string after the call
has been answered by the DID extension as an automatic
nd
2
dialing.
) If you set a DID extension in a trunk, then only that
extension can use this trunk to call out, and all incoming
calls to this trunk will connect to that extension directly.
When selected DID Bynumber, enter configurations in DID
Prefix and DID Stripping to have the incoming calls
directed to the corresponding extension derived by number
manipulation. The SIP trunk numbers is therefore regarded
as the direct line of the extension.
When selected DID By Privilege, select a usergroup i n the
list as the privilege of inbound calls from this trunk. Enter
configuration in DID Prefix and DID Stripping to have the
incoming calls redirected to dial out.
DID Prefix A digit string to be prefixed to the incoming called number
after stripping.
DID Stripping A number of leadin g digits to be stripped from the original
14
called number. Click All to strip all digits of the original called
number.
Centrex DID
IVR List
7
Select this function to make the numbers of
incoming/outgoing calls more flexible by prefixing/stripping
digits.
All-number
Digitmap
Default Number
Inbound
Manipulation
Digitmap
Outbound
Manipulation
Digitmap
Associate an IVR menu with incoming calls to this trunk.
This is mandatory unless the trunk is configured for DID.
Leave it blank and the system will automatically create an
Select a digitmap ID for calls via the
trunk changing numbers.
Enter digits for calls from this trunk
displaying this caller ID.
Select a digitmap ID to have the
incoming calls direct to the
corresponding extensions.
Select a digitmap ID to have the
outgoing calls display the
corresponding numbers.
IVR for the trunk.
Usergroup8 of Privilege When disabled DID, click a usergroup in the list whose
reachability to other usergroups and trunks will be used as
the privilege of inbound calls from this trunk.
) There may not be appropriate usergroups to select
initially. One can come back later once the expected
usergroup has been added.
Disable Fast Bridging Select to disable express media forwarding.
) With Fast Bridging feature enabled, if the two parties
involved in a call (for example, one IP extension and
one SIP trunk) use different DTMF modes
(RFC2833/SIP INFO/Inband), inline transfer (*#) or
7
Please refer to 7.12 for details.
8
Please refer to 6.1 for details.
15
2nd-dialing might fail. To avoid such problem, it is
recommended to set the same DTMF mode for all IP
extensions and SIP trunks in the IPBX, as well as for
all IP phones registered to the IPBX. If it is not
feasible to set the same DTMF mode for some IP
extensions or SIP trunks, and inline transfer or
2nd-dialing is necessary for those IP extensions or
SIP trunks, the Fast Bridging feature can be disabled
on a per IP extension and per SIP trunk basis. Note
that Fast Bridging is enabled by default.
3.2.4.2 Add Analog PSTN Tr unk s
1. Enter settings shown in Table 3-9.
2. Click ADD to see the newly added analog PSTN trunk in the table in the webpage.
For deleting an analog PSTN trunk, select a trunk identifier and click DEL.
Click Next to assign trunks to usergroups.
Table 3-9 Add Analog PSTN Trunk Settings
Field Description
Trunk Group ID number of this PSTN trunk group. A valid number ranges
from 1 to 31. It should not overlap with existing ISDN PSTN
trunk groups.
Trunk Type Select the port type, FXO or FXS. If selecting FXS, users
can see By Number and By Privilege in the DID of
Extension list, and be able to configure DID Prefix and DID
Stripping.
Trunk Ports Select one or more FXO or FXS ports for this Analog PSTN
trunk.
Description Arbitrary description information.
Port Selection
Click to search for an available port in the group. Rotating
means to force ports being selected in turns to even cost.
DID T ype DID means direct inward dialing (also called DDI in Europe).
Select a preferred type, None DID, Extension DID, and DID
by Privilege from the list and then enter configuration in DID
Prefix and DID Stripping to have the incoming calls
16
directed to the corresponding trunk.
None DID
When selected None DID, all incoming calls will enter IVR
system instead of directly dial to a specified extension.
Select a preferred IVR from IVR list for this trunk.
Extension DID
When selected Extension DID, select an extension in the
list to be an unconditional destination for incoming calls to
this trunk. If prefix or stripping has been given, the result of
digit manipulation is dialed in a DTMF string after the call
has been answered by the DID extension as an automatic
nd
2
dialing.
) If you set a DID extension in a trunk, then only that
extension can use this trunk to call out, and all incoming
calls to this trunk will connect to that extension directly.
DID By Privilege When selected DID By Privilege, select a usergroup in the
list as the privilege of inbound calls from this trunk. Enter
configuration in DID Prefix and DID S t ripping to have the
incoming calls redirected to dial out.
DID Prefix A digit string to be prefixed to the incoming called number
after stripping.
DID Stripping A number of leadin g digits to be stripped from the original
called number. Click All to strip all digits of the original called
number.
Language Preferred language for system instructions heard from the
trunk.
IVR List
9
Associate an IVR menu with incoming calls to this trunk.
This is mandatory unless the trunk is configured for DID.
Leave it blank and the system will automatically create an
IVR for the trunk.
Usergroup10 of Privilege When disabled DID, click a usergroup in the list whose
reachability to other usergroups and trunks will be used as
9
Please refer to 7.12 for details.
10
Please refer to 6.1 for details.
17
the privilege of inbound calls from this trunk.
) There may not be any appropriate usergroups to select
initially. One can come back later to revise it, once the
expected usergroups are added.
Caller ID Detection Select to detect the Caller ID calling from PSTN lines.
Answering by Battery Reversal
Detection
If selected, billable time will count from the call is answered.
) Please enable this function when Central Office (CO)
site provides battery reversal.
3.2.4.3 Add ISDN PSTN Trunks
1. Enter settings shown in Table 3-10.
2. Click ADD to see the newly added ISDN PSTN trunk in the table in the webpage.
For deleting an ISDN PSTN trunk, select a trunk identifier and click DEL.
Click Next to assign trunks to usergroups.
Table 3-10 Add ISDN PSTN Trunk Settings
Field Description
Trunk Group ID number of this ISDN trunk group. A valid number ranges
Trunk Channels
from 1 to 31. It should not overlap with existing Analog
PSTN trunk groups.
The Trunk Channels is the logical ran ge of the sum of B
and D channels. Each physical ISDN port occupies three
Trunk Ports, two B and on e D channels. User only needs to
specify the B channel number here, since D channel is
reserved in the 3
E.g. Assume there are four ISDN ports in the PBX and no
other FXO/FXS modules installed, then one can set each
pair of numbers here, like 1,2 but excluding 3,6,9,12.
rd
trunk port for each physical ISDN port.
) If a four-port FXO/FXS module is also installed, then
the Trunk Ports here shoul d be numbered from 5 to 16
instead of 1 to 12. Make sure to specify the indices of
ports correctly, or PBX will not start. One can refer to
18
the POTS Setting page before configuration.
Description Arbitrary description information.
Port Selection
DID T ype DID means direct inward dialing (also called DDI in Europe).
None DID
Extension DID
Select to search for an available port in the group. Rotating
means to force ports being selected in turns to even cost.
Select a preferred type, None DID, Extension DID, DID by NumberDID by Privilege or Centrex DID from the list and
then enter configuration in DID Prefix and DID Stripping to
have the incoming calls directed to the corresponding trunk.
When selected None DID, all incoming calls will enter IVR
system instead of directly dial to a specified extension.
Select a preferred IVR from IVR list for this trunk.
When selected Extension DID, select an extension in the
list to be an unconditional destination for incoming calls to
this trunk. If prefix or stripping has been given, the result of
digit manipulation is dialed in a DTMF string after the call
has been answered by the DID extension as an automatic
nd
2
dialing.
) If you set a DID extension in a trunk, then only that
extension can use this trunk to call out, and all incoming
calls to this trunk will connect to that extension directly.
DID By Number
DID By Privilege
DID Prefix A digit string to be prefixed to the incoming called number
DID Stripping A number of leadin g digits to be stripped from the original
When selected DID Bynumber, enter configurations in DID Prefix and DID Stripping to have the incoming calls
directed to the corresponding extension derived by number
manipulation. The SIP trunk numbers is therefore regarded
as the direct line of the extension.
When selected DID By Privilege, select a usergroup i n the
list as the privilege of inbound calls from this trunk. Enter
configuration in DID Prefix and DID Stripping to have the
incoming calls redirected to dial out.
after stripping.
called number. Click All to strip all digits of the original called
number.
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