APBX IP04 User Manual
For Firmware Version: V1.4.0
2013-07-11
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Contents
Contents ........................................................................................................................................................................... 1
1. Introduction ............................................................................................................................................................... 3
2. Connection and Change IP Address of APBX ............................................................................................................. 4
2.1 Connection ..................................................................................................................................................... 4
2.2 WAN Settings ................................................................................................................................................. 5
2.3 System Status ................................................................................................................................................. 5
3. Create local extensions and make interior calls ........................................................................................................ 6
3.1 FXS extensions ............................................................................................................................................... 6
3.2 SIP extensions ................................................................................................................................................ 9
3.3 Register onto APBX with your IP phone ....................................................................................................... 13
3.4 Extensions Status ......................................................................................................................................... 13
3.5 Feature Codes .............................................................................................................................................. 14
3.6 SMTP Settings .............................................................................................................................................. 16
3.7 Conference ................................................................................................................................................... 18
3.8 Paging / Intercom ......................................................................................................................................... 18
3.9 Options ......................................................................................................................................................... 18
4. Create SIP trunk and make outbound / inbound calls to / from VoIP provider ...................................................... 19
4.1 Create SIP trunks .......................................................................................................................................... 19
4.2 Check SIP Trunk Status ................................................................................................................................. 21
4.3 Make outbound calls.................................................................................................................................... 21
4.4 Make inbound calls ...................................................................................................................................... 22
5. Make outbound / inbound calls to / from PSTN network ....................................................................................... 24
5.1 Make sure FXO modules are installed ......................................................................................................... 24
5.2 Make outbound calls.................................................................................................................................... 24
5.3 Make inbound calls ...................................................................................................................................... 26
6. Inbound Call Control ................................................................................................................................................ 27
6.1 Time Interval ................................................................................................................................................ 27
6.2 Hunt / Ring Group ........................................................................................................................................ 27
6.3 Queue ........................................................................................................................................................... 28
6.4 Auto Attendant / Voice Menu / IVR ............................................................................................................. 31
6.5 DISA .............................................................................................................................................................. 32
6.6 Blacklist ........................................................................................................................................................ 33
6.7 SIP Settings ................................................................................................................................................... 33
7. Audios ...................................................................................................................................................................... 35
7.1 Music On Hold .............................................................................................................................................. 35
7.2 Custom Prompts .......................................................................................................................................... 35
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7.3 Language Setting .......................................................................................................................................... 35
8. Network Settings ..................................................................................................................................................... 36
8.1 Web Access .................................................................................................................................................. 36
8.2 Firewall ......................................................................................................................................................... 36
8.3 Port Forwarding ........................................................................................................................................... 39
8.4 DDNS ............................................................................................................................................................ 39
8.5 VLAN ............................................................................................................................................................. 39
9. System Settings ....................................................................................................................................................... 40
9.1 Change Password ......................................................................................................................................... 40
9.2 Date && Time .............................................................................................................................................. 40
9.3 Firmware Upgrade ....................................................................................................................................... 41
9.4 Backup and Restore ..................................................................................................................................... 42
9.5 Reboot && Reset ......................................................................................................................................... 43
10. Reports .................................................................................................................................................................... 43
10.1 Call Detail Records ....................................................................................................................................... 43
11. Web Interface for extension.................................................................................................................................... 44
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1. Introduction
Overview of the APBX
ATCOM IPPBX is a SIP-based IP voice switch with a small embedded OS and rich GUI (Graphical User Interface),
providing a powerful networking and corporate communication function. With it, users can quickly deploy an internal
communication system for enterprise, as well as configure conveniently applications and value-added services on IP
PBX via its GUI, to fit enterprise's own various demands.
Targeting for SOHO user and SMB market with an easy to use graphical interface, IP04 provides a cost-saving
solution on their telecommunication data needs. With IP04, company with branch offices in different countries can be
easily combined together to work like a virtual single office through internet.
Hardware Specifications
CPU
400MHz Blackfin 532 Chip
NAND Flash
256 M
SDRAM
128 M
Analog Port
4
Network Interface
WAN
Measurement and Weight
Inner box
225 * 120 * 30 mm
G.W./unit
0.765KG
Carton MEAS
456 * 442 * 362 mm
Units per Carton
21 units/ CTN
G.W./CTN
18 KG/CTN
Function Features
Voicemail
Authentication before call outbound
Voicemail to Email
User WEB portal
Blind/Attended Transfer
Blacklist
Call Forward
Call Detail Records(CDR)
Call Parking
Conference
Do not Disturb (DND)
Ring Group
Group / Directed Pickup
Call Queue
Call Recording
IVR
Call Waiting
Intercom/Paging
Call Routing
Firewalls
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Caller ID
IP Restriction
BLF Support
DDNS
Music on Hold
VLAN
Storage Quota Privilege
2. Connection and Change IP Address of APBX
2.1 Connection
The default IP address of APBX is:
WAN: 192.168.1.100/255.255.255.0
The network scenario should be like below:
Ext:6001
192.168.1.3
Ext:6002
192.168.1.4
Ext:6003
192.168.1.5
Ext:6004
192.168.1.6
Ext:6005
192.168.1.7
WAN
192.168.1.100
1) Connect APBX to your PC directly or through switch.
Make sure IP address of your PC is in network 192.168.1.0/255.255.255.0, if not, you need to appoint an IP
address for your PC, for example, 192.168.1.3
2) Login APBX as administrator via WEB GUI
User: admin
Password: atcom
3) Go to Network Settings ->WAN.
Re-set IP of WAN port.
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2.2 WAN Settings
There are two ways to set an IP address to WAN port: DHCP, Static IP.
DHCP
APBX will obtain an IP address automatically from DHCP server when rebooting. It’s not recommended to choose
this option unless there is a reserved IP for APBX in DHCP server so that APBX can keep the same IP all the time.
Static IP
Set an IP address manually according to the real network environment. If APBX is behind a router, the gateway is
usually set to the IP of the router.
2.3 System Status
1) General
Product Model
Show the model of this APBX
Firmware Version
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Show the firmware version
System Uptime
Show the time how long the system has been running
System Current Time
Show the current time
2) Network
Show the network setting of APBX
3) Peripheral
Show what kinds of / how many modules are detected.
3. Create local extensions and make interior calls
There are two kinds of extensions in APBX: FXS extensions and SIP extensions.
3.1 FXS extensions
It needs support of FXS module, the module installed in IP04 can be: AX110S
Analog phone is available to make calls once connected to the corresponding FXS port, APBX configures FXS
extension automatically when FXS module is detected. FXS extension can’t be deleted. The extension number is
defined in PBX Settings -> Options -> Extension Preference, changing it can change the FXS extension number.
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1) General
Extension
Extension number, i.e. 6101, it is associated with this particular user / phone.
Port
The analog port bound with extension.
Name
A character-based name for this extension, i.e. ‘Bob Jones’.
Caller ID
CID showed in the other’s phone during a call, default is Extension.
2) Voice Mail
Enable Voice Mail
Check this option to enable voice mail account for the extension. Enabled by default.
Voice Mail Access PIN Code
Password for accessing this voice mail account, default is 123456. It’s also the password for extension to login his
administration web page.
3) Mail Setting
Enable Sending Voice Mail
Check this option to enable PBX send new voicemail to Email address below as an attachment.
Email Address
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The Email address that new voicemail will be send to when Enable Sending Voice Mail is enabled and PBX settings
-> SMTP settings is right set.
4) Flash
Hook Flash Detection Time
Sets the amount of time, in milliseconds, that the hook-flash must remain depressed in order for asterisk to consider
such an event a valid flash event. The default value of it is 1250 ms and it can be configured in 1 ms increments.
Sequential Hook Flash Interval
Sets the amount of time, in milliseconds, that must have passed since the last hook-flash event received by asterisk
before it will recognize a second event. If a second event occurs in less time than defined in here, then asterisk will
ignore the event. The default value is 750 ms, and it can be configured in 1 ms increments.
5) Follow Me
Follow me is a feature to let an incoming call to a called party to be redirected to a third party, the third party can be a
voicemail box, ring group, mobile telephone and so on.
When callee is No Answer / Busy / Unreachable, incoming calls will go to voicemail by default, if voicemail is
disabled, call will be hung up.
6) Other Options
Pickup Group
Allows extension to answer someone else’s telephone call by dialing the group call pickup code (defined in PBX
Settings->Feature Codes->General), the two extensions must be in a same pickup group.
APBX supports 10 pickup groups: 0-10, None means the extension belongs to none pickup group, extensions in group
None can’t pick up others’ ring call and also can’t be picked up by others.
Call Waiting
Check this option to enable the Call Waiting capability for this extension. Then the extension can answer a new call
when it is already on the line. If this Option is checked, the follow me option “When busy” will be unavailable.
Ring Out
Set the ring timeout for this extension. APBX will stop ringing the extension if the time is up and there is still no
answer.
Use Web Interface
When checked, user can login the administration web page of this extension with extension number and voice mail pin
code as username and password.
Storage Quota Privilege
Set capacity of disk space for this extension to store voicemail and call recording.
Restricted: 1 M
Basic: 2 M
Regular: 3 M
Privileged: 4 M
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Super: 5 M
3.2 SIP extensions
SIP extension is an SIP account that allows IP phone or softphone to register to. It can be created / modified / deleted
one by one or in batch.
Click to add an extension or to add multiple extensions. The
extension number range is defined in PBX Settings -> Options -> Extension Preference, changing it can create
extensions in others number range.
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1) General
Name
A character-based name for this extension, i.e. ‘Bob Jones’
Extension
Extension number, i.e. 6000, it is associated with this particular user / phone.
Password
Authentication for SIP phone to register and make calls.
Caller ID
CID showed in the other’s phone during a call, default is Extension.
Transport
The transplant protocol type for VoIP data package, default is UDP. Please make sure TCP is enabled in PBX Settings
-> SIP Settings before using TCP.
2) Voice Mail
Enable Voice Mail
Check this option to enable voice mail account for the extension. Enabled by default.
Voice Mail Access PIN Code
Password for accessing this voice mail account, default is the extension number. It’s also the password for extension
to login his administration web page.
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3) Mail Setting
Enable Sending Voice Mail
Check this option to enable PBX send new voicemail to Email address below as an attachment.
Email Address
The Email address that new voicemail will be send to when Enable Sending Voice Mail is enabled and PBX settings
-> SMTP settings is right set.
4) Follow Me
Follow me is a feature to let an incoming call to a called party to be redirected to a third party, the third party can be a
voicemail box, ring group, mobile telephone and so on.
When callee is No Answer / Busy / Unreachable, incoming calls will go to voicemail by default, if voicemail is
disabled, call will be hung up.
5) Other Options
Pickup Group
Allows extension to answer someone else’s telephone call by dialing the group call pickup code (defined in PBX
Settings->Feature Codes->General), the two extensions must be in a same pickup group.
APBX supports 10 pickup groups: 0-10, None means the extension belongs to none pickup group, extensions in group
None can’t pick up others’ ring call and also can’t be picked up by others.
Call Waiting
Check this option to enable the Call Waiting capability for this extension. Then the extension can answer a new call
when it is already on the line. It also needs the call waiting support of IP phone. If this Option is checked, the follow
me option “When busy” will be unavailable.
Ring Out
Set the ring timeout for this extension. APBX will stop ringing the extension if the time is up and there is still no
answer.
Use Web Interface
When checked, user can login the administration web page of this extension with extension number and voice mail pin
code as username and password.
Storage Quota Privilege
Set capacity of disk space for this extension to store voicemail and call recording.
Restricted: 1 M
Basic: 2 M
Regular: 3 M
Privileged: 4 M
Super: 5 M
6) Advanced Configuration
SIP Settings
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NAT
Try this setting when APBX is on a public IP, communicating with devices hidden behind a NAT device (broadband
router). If you have one-way audio problems, you usually have problems with your NAT configuration or your
firewall’s support of SIP+RTP ports.
Can Reinvite
By default, Asterisk will route the media streams from SIP endpoints through itself. Enabling this option causes
asterisk to attempt to negotiate the endpoints to route the media stream directly, bypassing asterisk. It is not always
possible for asterisk to negotiate endpoint-to-endpoint media routing.
DTMF Mode
Select DTMF sending mode, there are three modes: rfc2833, inband, info. The DTMF setting in here should be as
same as that in SIP phone, otherwise APBX will not detect the users’ input correctly during a call.
Auto means IPPBX will match anyone of them according to the setting of SIP phone.
Preferred Codec
Set the allowed codec and priority for SIP phone. The options are below:
A-law, U-law, GSM, SPEEX, G726, G722, ADPCM, G729
IP Restriction
Enable it to permit trusted IP/network register to this extension number. This is an useful way to improve the security
of APBX.
NOTE: There must be at least one same codec chose in APBX extension settings and SIP phone codec
settings, otherwise, It’s impossible to make calls between APBX and SIP phone.
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Set trusted IP: xx.xx.xx.xx/255.255.255.255, for example: 192.168.1.160/255.255.255.255
Set trusted network: xx.xx.xx.xx/subnet mask, for example: 192.168.1.0/255.255.255.0
3.3 Register onto APBX with your IP phone
After successfully register with 6000 and 6001, you can make interior calls among 6000, 6001, 6101(FXS), 6102(FXS)
now.
3.4 Extensions Status
This page is used to check the extensions status.
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Idle: The extension is registered and idle.
Busy: The extension is on the phone.
Ringing: The extension is ringing.
Unavailable: The extension is not registered and unreachable.
3.5 Feature Codes
1) General
Call Recording
Record a call while in the call.
Dial Call Recording Code to begin recording and dial it again to stop recording during a call.
Checking Voicemail
Users can check their Voicemail by dialing this code on their phone.
Attended Transfer
Routed a call to a third party only if the third party answers the call. The call flow should be like below:
1. Phone A call B, B answers the call.
2. B presses feature code(*3) and C’s number to transfer the call to C
3. If C answers B’s call, B can talk to C and A is on hold
4. If Phone B hangs up, A will talk to C, transfer is successful.
4’ If Phone C hangs up, B connects back to A, transfer is failed
Blind Transfer
Blind transfer is when a call is routed to a third party, the original call is ended, and no check is made to determine
whether the transferred call is answered or if the number is busy.
The call flow should be like below:
1. Phone A call B, B answers the call.
2. B presses feature code(*03) and C’s number to transfer the call to C
Group Call Pickup
Pick up a ring call for other extensions in the same pickup group.
The call flow should be like below:
1. C calls A, phone A is ring, but A is not at his/her seat.
2. Extension A and B are in the same pick up group, B can dial Group Call Pickup code to pick up the ring call, and
talk to C.
Direct Call Pickup
Pick up a ring call for an appointed extension.
If this page response slowly, please be patient to wait the output before check other pages. Otherwise, other pages
cannot be displayed correctly since APBX is accessing database while status checking, and database is locked for
other pages’ request.
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The call flow should be like below:
1. C calls A, phone A is ring, but A is not at his/her seat.
2. B can dial Direct Call Pickup code + A’s extension number to pick up the ring call, and talk to C.
Intercom
Connect directly to a specified phone.
The call flow should be like below:
1. A dial Intercom code + B’s extension nubmer.
2. If Phone B supports page/intercom, it will answer the call automatically.
2) Call Park
It allows a person to park a call on IPPBX and continue the conversation from any other telephone set.
The call flow should be like below:
1. A and B are on the conversation.
2. A dial call park code (e.g. *6), PBX will tell A a park extension (e.g. 701) and then hang up the call. B is parked
on PBX.
3. C dial park extension: 701, PBX will bridge C and B.
3) Call Forward
Users can configure their follow me settings via their phones.
Reset to Defaults
Reset follow me settings by dialing *70 (default code, can be changed). After dialing in, PBX will prompt a “beep”,
then the setting is completed and the call will be hung up.
Enable/Cancel Unconditional Call Forward
Enable/Disable call forward Always function.
Enable/Cancel Call Forward On Busy
Enable/Disable call forward When Busy function.
Enable/Cancel Call Forward On No-Answer
Enable/Disable call forward When No Answer function.
Call Forward to Number
Set the destination for call forward to number by dialing *74 (default code, can be changed), if the number is not set
yet, dial *74+number to set it.
Call Forward to Voice Mail
Set the destination for call forward to voicemail.
NOTE: Default Follow Me settings are as below:
Always: Disabled
When no answer: Enabled
When busy: Enabled
Forward to: Voice Mail
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Enable/Cancel Do Not Disturb
Enable/Disable Do Not Disturb function
3.6 SMTP Settings
1) Voice Mail to Email Setting
Email Address
The Sender Email Address APBX used to send voicemail.
Password
The password for above Email Address/Account.
SMTP Server
SMTP server that above email address/account is located in.
Port
Port for SMTP server, for example: Gmail server use port 465 to send / receive email.
Use SSL/TLS to send secure message to server
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Some servers need to authenticate sender before sending email, then the box should be checked.
Test SMTP Settings
Check whether the SMTP setup is OK. PBX will send an email to the test email address using above SMTP setting
information. If the test failed, please check that information and network connection.
2) Voice Mail Setting
Max Messages
This limits the number of messages in a voicemail folder. The maximum value is 9999 (hard coded) and the default
100. When a mailbox has more than this number of messages in it, new messages can not be recorded and “voice mail
box is full” is played to the caller.
Max Messages Time
This defines the maximum amount of time in seconds of an incoming message. Use this when there are many users
and disk space is limited. The default value is 120 (2 minutes), 0 means there will be no maximum time limit
enforced.
Min Messages Time
This setting can be used to eliminate messages which are shorter than a given amount of time in seconds. The default
value for this setting is 5.
Say CID/Duration
Read back caller’s telephone number / message duration prior to playing the incoming message when checking it.
Envelope
Envelope controls whether or not Asterisk will play the message envelope (date/time) before playing the voicemail
message.
Review
Let a caller review their message before committing it to a mailbox.
NOTE: After SMTP setting, please set Email address for each extension to achieve Voicemail to Email
function.
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Ask Caller to Dial 5
If this option is set, the caller will be prompted to press 5 before leaving a message.
3.7 Conference
Allows participants dial into a virtual meeting room from their own phone, support up to 20 participants.
Conference Room
Extension number of conference room, participant dial it to get into the room.
PIN#
Used for authentication before participants dial into the room, APBX will playback MoH for the first participant.
3.8 Paging / Intercom
Dial a code and connect directly to a built-in two-way announcement and talkback function on one or more phones,
support up to 20 participants.
Paging Group Number
Extension number of paging group, dial it to reach this group.
Duplex
If checked, caller and callees all can speak and hear. Otherwise, only caller can speak, and callees can hear.
3.9 Options
1) General Preference
Ring Timeout
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Default Ring Timeout for an extension if Ring Out for it is not set.
Max Call Duration
This defines the maximum amount of time in seconds for a interior call, 0 means no limit, default is 6000s.
Music On Hold
This define which Music on hold is used when transfer/call park/on hold/Conference etc.
Tone Region
This defines how the default dial tone, busy tone, and ring tone look like, please select your country or nearest
neighboring country here.
2) Extension Preference.
Defines the range for SIP / FXS / Ring Group / Voice Menu / Paging Group / Conference / Queue Extensions. The
extension length must be between 3 and 9 digits. The maximum quantity can be supported for each are as below:
SIP extension
100
FXS extension
8
Ring Group
9
Voice menu/IVR
16
Paging Group
9
Conference
9
Queue
9
4. Create SIP trunk and make outbound / inbound calls to / from VoIP provider
4.1 Create SIP trunks
Go to PBX Settings -> Trunks, Click to add a new SIP trunk.
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Trunk Name
A unique label to help you identify the trunk.
Provider Hostname/IP
Hostname or IP of your VoIP provider, default port is 5060
Account Name
The username that your service provider configured
Authuser
The username that your service provider configured for authentication, generally, it’s same as Account Name.
Password
The password configured for the user in your service provider side.
Enable Outbound Proxy
Outbound Proxy is a SIP proxy server, it acts, like any proxy server, as a middleman between two communicating
agents, serving as a transit point for all SIP traffic. It can be used to solve the SIP one-way-audio issue.
Outbound Caller ID
The Caller ID used when using outbound proxy.
SIP Transport
The transplant protocol type for VoIP data package, default is UDP. Please make sure TCP is enabled in PBX Settings
-> SIP Settings before using TCP.
Maximum Outbound Calls
Define the maximum quantity of outbound connections (simultaneous calls) that can be used on this trunk. Inbound
calls are not counted in. 0 means no connection limit.
Preferred Code
Set the allowed codec and priority for this trunk.
Advance Configuration
DOD(Direct Outward Dialing Number) Setting
Set the Outbound number for different extensions.
For example:
< === Do not set DOD,
The other end of the trunk will
show original (interior) extension
number.
DOD is set ==== >
The other end of the trunk will
show DOD number.
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4.2 Check SIP Trunk Status
After creating trunk, go to System Status -> Trunk Status to check the SIP trunk Status, make sure it’s registered.
4.3 Make outbound calls
Go to PBX Settings -> Outbound Routes, click to add an outbound route.
Outbound Route Name
A unique label to help you identify the outbound route.
Dial Pattern
A filter for marching numbers you dial, the call will be forwarded out via Selected Trunks only when it matches the
dial pattern here. In patterns, some characters have special meanings.
X means Any Digits from 0-9
Z means Any Digits from 1-9
N means Any Digits from 2-9
[1234-9] means Any Digits in the brackets (in this example, 1, 2, 3, 4, 5, 6, 7, 8, 9)
. means one or more digits
! will match none remaining digits, and causes the matching process to complete as soon as it can be determined that
no other matches are possible.
For example: Once set Dial Pattern: 2XX Strip: 0, that means any calls to 200-299 will be forwarded out.
Please do not simply set it to X., otherwise all telephone numbers with 2 or 2+ digits will be matched, this
outbound route probably affect your interior calls, unless your local extensions is just a single figure.
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Strip: The number of digits that will be stripped from the front of the dialing string before the call is placed via
Selected Trunks. See example in Chapter 5.
Prepend these digits: Allows the user to specify digits that are prepended before the call is placed via the trunk. See
example in Part 4.
Password: Authentication for Selected Extensions before dialing out.
Outbound Extension Selection: Select extensions which can dial out with this outbound route. In my case, only 6000
and 6001 can dial out with this trunk.
Outbound Trunk Selection: Select trunks which calls are forwarded out through.
4.4 Make inbound calls
Go to PBX Settings -> Inbound Routes, click to add an inbound route.
Caller ID
Define the Caller ID number to be matched on incoming calls. Leave this field blank to match any or no CID info.
Special characters described in chapter 4.3 can be used here as same.
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DID number
Define the expected DID number if your trunk passes DID on incoming calls. Leave this blank to match calls with any
no DID info. Special characters described in chapter 4.3 can be used here as same.
Extension
Define the extension for DID number. This field is only valid when you use SIP trunk for this inbound router. You can
only input number and ‘-’ in this field, and the format can be XXX or XXX-XXX. The count of the number must be
only one or equal the count of the DID number. Up to 100 DID numbers can be set.
Inbound Trunk Selection
Select the trunks for which this inbound route apply.
Time
Select appropriate time intervals for when this inbound route apply.
For example: Set DID number: 6000-6010，Extension: 6000，All inbound calls to 6000-6010 will be forwarded
to extension 6000. Set DID number: 6000-6010, Extension: 6000-6010, inbound calls to 6000-6010 will be
forwarded to corresponding extension.
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Path
Set the destination for incoming calls. If Extension is set, this option will not take effect.
5. Make outbound / inbound calls to / from PSTN network
5.1 Make sure FXO modules are installed
If there are FXO modules installed in your APBX, APBX configures analog trunk automatically when they are
detected. The module installed in IP04 can be: AX110X
5.2 Make outbound calls
Go to PBX Settings -> Outbound Routes, click to add an outbound route.
Outbound Route Name
A unique label to help you identify the outbound route.
Dial Pattern
NOTE: Before using them, please make sure FXO port is connected with PSTN line (InService). The
connection status can be checked in System Status -> Trunk Status
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A filter for marching numbers you dial, the call will be forwarded out via Selected Trunks only when it matches the
dial pattern here. In patterns, some characters have special meanings.
X means Any Digits from 0-9
Z means Any Digits from 1-9
N means Any Digits from 2-9
[1234-9] means Any Digits in the brackets (in this example, 1, 2, 3, 4, 5, 6, 7, 8, 9)
. means one or more digits
Strip: The number of digits that will be stripped from the front of the dialing string before the call is placed via
Selected Trunks.
Prepend these digits
Allows the user to specify digits that are prepended before the call is placed via the trunk.
Password
For example: If set Dial Pattern: 9, Strip: 1, Prepend 123, user need to dial 94567 to dial PSTN number
1234567
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Authentication for Selected Extensions before dialing out.
Outbound Extension Selection
Select extensions which can dial out with this outbound route. In my case, only 6000 and 6001 can prefix 9 to dial out.
Outbound Trunk Selection
Select trunks which calls are forwarded out through. In my case, the call will be forwarded out via analog trunk FXO3
first, if failed, will try FXO4.
5.3 Make inbound calls
Go to PBX Settings -> Inbound Routes, click to add an inbound route. Just setting Selected Trunks and
Path is OK.
Caller ID
Define the Caller ID number to be matched on incoming calls. Leave this field blank to match any or no CID info.
Special characters described in chapter 5.2 can be used here as same.
DID number
Just leave it blank.
Extension
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It’s unavailable for Analog trunk, leave it blank.
Inbound Trunk Selection
Select the trunks for which this inbound route apply.
Time
Select appropriate time intervals for when this inbound route apply.
Path
Set the destination for incoming calls.
6. Inbound Call Control
6.1 Time Interval
Set the Time Interval for inbound route.
6.2 Hunt / Ring Group
This defines a 'virtual' extension that rings a group of phones simultaneously / one by one, stopping until any one of
them is picked up.
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6.3 Queue
Usually used in Call Centre to queue customers for the next available operator.
1) General
Queue Name
Name of the queue
Queue Number
Extension number of the queue, dial it to get into the queue
Queue Password
Used as authentication for users before being dynamic agent.
Queue Agent Timeout
Ring timeout in seconds when calling an agent
Queue Max Wait Time
The maximum time in seconds for a caller can wait in the queue before being pulled out. (0 means unlimited)
Queue Ringing Strategy
Strategy for APBX ring the agents.
RingAll: Ring all available agents simultaneously until one answers.
LeastRecent: Ring agents which was least recently called.
FewestCalls: Ring agents with the fewest completed calls.
Random: Ring agents in a random way.
RRmemory: Round robin with memory, remembers where it left off in the last ring pass.
2) Agents
Select Static Agent here. there are two kinds of agents:
Static Agent: chose here
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Dynamic Agent: users can dial ‘Queue number + *’ to log in as dynamic agent, and ‘Queue number + **’ to log out.
In this case, users can dial 6900* to being a dynamic agent (need to enter password 123), and 6900** to log out.
3) Caller Position Announcement
Announce queue position and / or estimated hold time to caller
4) Period Announcement
This allows a message like "Thank you for holding, your call is important to us." to be played at regular intervals
while a caller is in the queue
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5) Event
This allows callers waiting in the queue to dial a key to go to other destination.
6) Failover Destination
This define the failover destination for callers when the max wait time is up.
7) Others
Music On Hold
Select Music On Hold Class for this Queue
Leave When Empty
This option controls whether calls already on hold are forced out of a queue that has no agents. There are two options:
Yes: Callers are forced out of a queue when no agents logged in, or if all logged in agents are unavailable.
NO: Callers will remain in a queue with no agents.
Join Empty
This option controls whether callers can join a call queue that has no agents. There are three options:
Yes: Callers can join a call queue with no agents or only unavailable agents.
No: Callers cannot join a queue with no agents or if all agents are unavailable.
Agent Announcement
Announcement played to the agent prior to bridging in the caller.
Join Announcement
Announcement played to callers once prior to joining the queue.
NOTE: The key point with announcements is that they are only played within the timeout/retry period set on the
queue. For the most part this works OK as when all queue members are busy/unavailable, the timeout/retry period
is effectively ignored (i.e. you can consider the queue to always be in this state) and announcements will be played
as per your setting of the announce-frequency and periodic-announce-frequency parameters. When a handset is
available and the queue is ringing it, the timeout/retry timeouts become critical. For example, if you want
announcements every 20 seconds, but the timeout is set to 60 seconds, when a queue member is ringing, you will
only ever get announcements every 60 seconds.
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Retry
How long does APBX wait before trying all the members again.
Wrap Up Time
After a successful call, how long to wait before sending a potentially free member another call.
6.4 Auto Attendant / Voice Menu / IVR
Callers are presented with a recorded menu and respond by selecting a digit or, in some cases, by entering an
extension number. The automated attendant eliminates the need for a live operator to handle the call.
Voice Menu Name
Name of the Voice Menu
Voice Menu Number
Extension number of the voice menu, dial it to get into the voice menu
Key Timeout
How long for APBX to wait user’s input
Repeat Count
How many times to play prompt
Allow Dialing of Other Extensions
Allow dialing local extensions
Key Press Event
Dial digit to trigger corresponding event
No Entry Forward to
The destination for incoming call if there is none input
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Invalid Forward to
The destination for incoming call if there is invalid input
6.5 DISA
DISA (Direct Inward System Access) allows someone calling in from outside to obtain an "internal" system dialtone
and dial out as if a local extension.
1) General
DISA Name
A name for the DISA
PIN #
When caller get into the DISA, this password is needed to put before making calls.
Response Timeout
The maximum time in seconds APBX will wait for input from a user.
Digit Timeout
The maximum time allowed between entry of digits. If exceeded, user input is deemed to have finished.
2) Outbound Trunks
Choose the outbound route that callers can use to dial out.
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For example:
Both City A and B have a APBX, APBX-A and APBX-B, they are connected with SIP trunk, and APBX-A has FXO
trunk to connect local PSTN and outbound route for that, DISA can be used as below:
1. Create a DISA in APBX-A including the FXO trunk.
2. Set it as the destination of inbound route for SIP trunk.
After users of APBX-B dial into DISA application in APBX-A, The DISA application in turn requires the user to
enter his passcode, followed by the pound sign (#). If the passcode is correct, the user will hear dialtone on which a
outbound call may be placed, so there is no long distance call fees.
6.6 Blacklist
Block incoming calls from specified numbers
If a number in blacklist dial into APBX, caller will hear following prompt: “Then number you have dialed is not in
service. Please check the number and try again.” Then system will then disconnect the call.
6.7 SIP Settings
1) General
UDP Port
Set the SIP port (UDP) which APBX is listening to.
Enable TCP
Enable TCP protocol for SIP.
TCP Port
Set the SIP port (TCP) which APBX is listening to.
Registration / Subscription Time Max
Maximum duration in seconds of a SIP registration / subscription.
Registration / Subscription Time Min
Minimum duration in seconds of a SIP registration / subscription.
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RTP Port Min / Max
Set the RTP port range.
DTMF Mode
Set the default DTMF mode
2) NAT
Here provide other two solutions for SIP one-way-audio issue besides outbound proxy. Using any one is OK.
a. STUN
Just setting STUN server / port is OK.
There are many public STUN server on Internet: http://www.voip-info.org/wiki/view/STUN
b. NAT
The External IP, External Host and Local Network Identification settings are used if you use APBX behind a NAT
device to communicate with services on the outside.
External IP address
Address that we're going to put in outbound SIP messages if we're behind a NAT. The externip and localnet is used
when registering and communicating with other proxies that we're registered with.
External Host
Alternatively you can specify an external host, and APBX will perform DNS queries periodically. Not recommended
for production environments! Use External IP instead.
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External Refresh Interval
How often to refresh External Host if used.
NAT Mode
Global NAT settings (Affects all peers and users), is used when Asterisk is on a public IP, communicating with
devices hidden behind a NAT device (broadband router). If you have one-way audio problems, you usually have
problems with your NAT configuration or your firewall's support of SIP+RTP ports.
Local Network identification
You may add multiple local networks. A reasonable set of defaults are set here.
Allow RTP Reinvite
By default, Asterisk tries to re-invite the audio to an optimal path. If there's no reason for APBX to stay in the media
path, the media will be redirected. This does not really work with in the case where APBX is outside and have clients
on the inside of a NAT. In that case, you want to set this option to nonat.
7. Audios
7.1 Music On Hold
Manage audio files for Music On Hold, the format should be .WAV and .GSM, the size should less than 4 MB.
7.2 Custom Prompts
Manage prompts used for Voice Menu. It can be recorded by extensions or uploaded from local PC.
7.3 Language Setting
Set the language of default system prompt audio, English is supported by default. French and Spanish need to be
download from Internet when chose at the first time. Make sure gateway is right set so that APBX can access Internet.
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8. Network Settings
Description of LAN, WAN and DHCP server settings can be found in Chapter 2. All network settings will take effect
after APBX reboot.
8.1 Web Access
Choose the web access protocol and port for web server here. HTTP and HTTPs are both supported, default port is 80
and 443 respective.
8.2 Firewall
Firewall is used to prevent unauthorized connections.
1) Enable Firewall
Check it to enable firewall.
2) Common Rule
Accept/Drop the connections from remote hosts.
Name
A name for the rule.
Description
Simple description for the rule.
Protocol
Set the protocol type for connection.
Port
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Set the destination port range for connection. The main protocols and default ports APBX uses for each application are
list below:
HTTP
TCP:80
HTTPS
TCP:443
SIP
UDP:5060
SIP
TCP:5060
RTP
UDP:50000-60000
IP
Set source IP of connection.
Format of IP: IP/mask
For example:
192.168.1.156/255.255.255.255 for IP 192.168.1.156
216.207.245.47/255.255.255.255 for IP 216.207.245.47
192.168.1.156/255.255.255.0 for network 192.168.1.0/24
Mac Address
Set source Mac of connection. Either IP or Mac Address must be set.
Action
Accept: Accept the access from remote hosts.
Drop: Drop the access from remote hosts.
3) Auto Defense
Limit connections from remote hosts.
Port
Set the destination port range for connection.
Protocol
Set the protocol type for connection.
Rate
The maximum packets or connections can be handled per second
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4) SIP Defense:
Limit connections to SIP port from remote hosts.
Port
Set the destination port range for connection.
Protocol
Set the protocol type for connection.
SIP Packets
The maximum packets can be handled per time interval.
Time Interval
Time unit which IPPBX uses to manage IP packets received.
5) Other Options
Disable Ping
Check this to drop ping packets from remote hosts.
Drop All
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Check this to drop all packets or connection from other hosts if there are no other rules defined.
8.3 Port Forwarding
When APBX works as a router, it can forward connections to WAN to a device connected to LAN network.
For example: user can access 192.168.10.2:80 (connected to LAN) by accessing xx.xx.xx.xx:8080 (xx.xx.xx.xx is the
IP of WAN)
8.4 DDNS
Dynamic Domain Name Service (DDNS) is a service used to map a domain name to the dynamic IP address of a
network device. IPPBX support 3 DDNS servers below, please go to the website of below servers and apply a domain
name then fill related information here.
dyndns.org
qdns
www.zoneedit.com
8.5 VLAN
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A VLAN (Virtual LAN) is a logical local area network (LAN) that extends beyond a single traditional LAN to a group
of LAN segments, given specific configurations. Both WAN and LAN support 2 VLANs.
9. System Settings
9.1 Change Password
Change the password for admin login, it will take effect immediately.
9.2 Date && Time
Set the date and time for APBX. The settings will take effect immediately.
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1) General
There are two ways to set Date/Time for APBX:
a. NTP server
Make sure the connection between APBX and NTP server is OK, if the NTP server is located on Internet, the
gateway of WAN should be right set so that APBX can access Internet.
GMT TimeZone is also an important arguments for time setting in this way. Please choose the right Time Zone.
b. Manually
2) Daylight Saving Time
There are two ways to set DST:
a. Automatic
Just making sure GMT Time Zone is right set is OK. There have already DST setting in each Time Zone.
b. Manually
However, the DST in some countries is changing every year. If the DST setting in Time Zone is not exact. Please
set it manually, the format should be: start=start_time;end=end_time;save=offset
The rule for start / end time is: month/mday/wday/hour:min:second
1<= month <=12 , 0< mday <=31 , 0<= wday <7
month/mday/wday means the first wday coming after month/mday
for example:
start=4/1/7/0:0:0;end=10/31/7/0:0:0;save=1 means APBX time from the first Sunday coming after April 1th to the
first Sunday coming after October 31th will be one hour early.
9.3 Firmware Upgrade
There are two ways to upload Kernel / Application for APBX:
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1) HTTP
Upload them from local PC.
2) TFTP
Upload them from TFTP server, the Kernel / Application must be located in base directory of TFTP server.
Choose Reset Config will reset the configuration.
9.4 Backup and Restore
1) Backup
Create Backup for configuration / System audio prompt / Voice Mail. The backup can be downloaded to local PC.
2) Restore
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Click Restore to restore corresponding backup, backup file can be uploaded from local PC. Backup will be used after
APBX reboot. It can’t be used for different product models.
9.5 Reboot && Reset
Reboot or Reset APBX.
10. Reports
10.1 Call Detail Records
Display the Call Detail Records, the operation for it can be search, delete and download.
1) Search
Users can search the records they needs according to Source, Destination, and / or Time.
2) Delete
APBX supports two delete operation: delete selected CDR and delete all CDR.
3) Download
It can be download to local PC
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11. Web Interface for extension
PBX allows users to check their voicemail / CDR, and set personal settings.
1. Check Use Web Interface option in PBX Settings -> Extensions management settings to allow this extension to
login its own web interface.
2. Enter the IP of PBX in the browser.
3. Login with extension number / Voice Mail Access PIN Code as username / password
1) Voice Mail Checking
Users can listen / download / delete / move voice mail here.
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2) CDR Checking
Users can check their CDR here.
3) Personal Settings
Users can set voice mail / voice mail to email / follow me / ring timeout here.
--Finish--
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