ATCOM APBX IP01 User manual

APBX IP01 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, IP01 provides a cost-saving
solution on their telecommunication data needs. With IP01, 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
64 M
Analog Port
1
Network Interface
WAN
Measurement and Weight
Inner box
100 * 100 * 28 mm
G.W./unit
0.36KG
Carton MEAS
456 * 442 * 362 mm
Units per Carton
40 units/ CTN
G.W./CTN
17 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 IP01 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
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.

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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
Super: 5 M

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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.