IPPBX IP01 User Manual
For Firmware Version: V3.0
2013-12-11
IPPBX IP01 User Manual
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Contents
Contents........................................................................................................................................................................... 1
1. Introduction .............................................................................................................................................................. 4
2. Connection and Change IP Address of IPPBX .................................................................................................. 5
2.1 Connection ................................................................................................................................................... 5
2.2 WAN Settings ............................................................................................................................................... 6
2.3 System Status .............................................................................................................................................. 7
3. Create local extensions and make interior calls ................................................................................................ 7
3.1 FXS extensions............................................................................................................................................ 8
3.2 SIP extensions ........................................................................................................................................... 11
3.2.1 Add new extensions ........................................................................................................................ 11
3.2.2 Modify extensions .......................................................................................................................... 15
3.3 Register onto IPPBX with your IP phone ............................................................................................... 15
3.4 Extensions Status...................................................................................................................................... 16
3.5 Feature Codes ........................................................................................................................................... 16
3.6 SMTP Settings ........................................................................................................................................... 19
3.7 Conference ................................................................................................................................................. 20
3.8 Paging / Intercom ...................................................................................................................................... 21
3.9 Options ........................................................................................................................................................ 21
4. Create SIP trunk to communicate with VoIP provider ..................................................................................... 22
4.1 Create SIP trunks ...................................................................................................................................... 22
4.2 Check SIP Trunk Status ........................................................................................................................... 24
4.3 Make outbound calls ................................................................................................................................. 25
4.4 Make inbound calls ................................................................................................................................... 26
5. Create Service Provider to connect two IP PBXes .......................................................................................... 28
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5.1 Create Service Provider ........................................................................................................................... 28
5.2 Check Service Provider Status ............................................................................................................... 29
5.3 Make outbound calls ................................................................................................................................. 29
5.4 Make inbound calls ................................................................................................................................... 30
6. Make outbound/inbound calls to/from PSTN network ..................................................................................... 31
6.1 Make sure FXO modules are installed ................................................................................................... 31
6.2 Make outbound calls ................................................................................................................................. 32
6.3 Make inbound calls ................................................................................................................................... 33
7. Inbound Call Control............................................................................................................................................. 34
7.1 Time Interval ................................................................................................................................................ 34
7.2 Hunt / Ring Group ....................................................................................................................................... 35
7.3 Queue ........................................................................................................................................................... 36
7.4 IVR ................................................................................................................................................................ 38
7.5 DISA .............................................................................................................................................................. 40
7.6 Call Back ...................................................................................................................................................... 41
7.7 Inbound to Outbound .................................................................................................................................... 43
7.7.1 FXO trunk to SIP trunk ...................................................................................................................... 43
7.7.2 SIP trunk to FXO trunk .................................................................................................................... 44
7.8 Custom Dialplan ............................................................................................................................................. 45
7.9 Blacklist ........................................................................................................................................................ 45
7.10 SIP Settings .................................................................................................................................................. 46
8. Audios ..................................................................................................................................................................... 47
8.1 Music On Hold ............................................................................................................................................. 48
8.2 Custom Prompts ......................................................................................................................................... 48
8.3 Language Setting ........................................................................................................................................ 48
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9. Network Settings ................................................................................................................................................... 48
9.1 Web Access ................................................................................................................................................. 48
9.2 Firewall ......................................................................................................................................................... 49
9.3 DDNS ............................................................................................................................................................ 52
9.4 VLAN ............................................................................................................................................................. 53
9.5 VPN ............................................................................................................................................................... 53
10. System Settings ............................................................................................................................................ 53
10.1 Change Password .................................................................................................................................... 53
10.2 Auto Provisioning ................................................................................................................................... 54
10.2.1 Auto Provisioning .......................................................................................................................... 54
10.2.2 Configured Phones ....................................................................................................................... 55
10.2.3 Unconfigured Phones ................................................................................................................... 56
10.2.4 Upload Phone Configure .............................................................................................................. 57
10.3 Date && Time ............................................................................................................................................ 57
10.4 External Storage ....................................................................................................................................... 58
10.5 Firmware Upgrade .................................................................................................................................... 60
10.6 Backup and Restore ................................................................................................................................. 60
10.7 Reboot && Reset ...................................................................................................................................... 61
11. Reports ........................................................................................................................................................... 62
11.1 Call Detail Records ................................................................................................................................... 62
11.2 Syslog ...................................................................................................................................................... 63
12. Web Interface for extension......................................................................................................................... 63
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1. Introduction
Overview of the IPPBX
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 * 28mm
G.W./unit
0.36KG
Carton MEAS
456 * 442 * 362 mm
Units per Carton
21 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
Call Back
Call Waiting
IVR
Call Routing
Intercom/Paging
Caller ID
IP Restriction
BLF Support
Firewalls
Music on Hold
DDNS
Video Call
DHCP Server
PPPoE
VLAN
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Port Forward
VPN client
External Storage
Auto Provisioning
Storage Quota Privilege
Custom Dialplan
Auto call recording
Integrated built-in packet capture tools
2. Connection and Change IP Address of IPPBX
2.1 Connection
The default IP address of IPPBX is:
WAN: 192.168.1.100/255.255.255.0
The network scenario should be like below:
Internet
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 IPPBX 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 IPPBX as administrator via WEB GUI
User: admin
Password: atcom
NOTE: Language option will be accomplished in future versions, currently, only English is supported. So it’s not
selectable now.
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
IPPBX 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 IPPBX in DHCP server so that IPPBX can keep the same IP
all the time.
Static IP
Set an IP address manually according to the real network environment. If IPPBX is behind a router, the gateway
is usually set to the IP of the router.
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2.3 System Status
1) General
Product Model
Show the model of this IPPBX
Firmware Version
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 IPPBX
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 IPPBX: FXS extensions and SIP extensions.
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3.1 FXS extensions
It needs support of FXS module, the module installed in IP01 can be:
AX110S: single FXS port
Analog phone is available to make calls once connected to the corresponding FXS port, IPPBX configures FXS
extension automatically when FXS module is detected and they can’t be deleted.
NOTE: FXS extension number range is defined in PBX Settings -> Options -> Extension Preference, changing
it can change the FXS extension number.
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) Volume Settings
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Rxgain
Adjust the volume sent to FXS extension. There are 3 options: 20%,60%,100%.
Txgain
Adjust the volume sent out by FXS extension. There are 3 options: 20%,60%,100%.
5) Flash
Hook Flash Detection Time
Sets the amount of time, in milliseconds, that the hook-flash must remain depressed in order for IPPBX 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
IPPBX before it will recognize a second event. If a second event occurs in less time than defined in here, then
IPPBX will ignore the event. The default value is 750 ms, and it can be configured in 1 ms increments.
6) 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.
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7) 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.
IPPBX 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. IPPBX will stop ringing the extension if the time is up and there is still no
answer.
Use Web Interface
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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 in IPPBX.
Restricted: 1 M
Basic: 2 M
Regular: 3 M
Privileged: 4 M
Super: 5 M
NOTE: If you set System Settings -> External Storage and set Move Files Created Before 0 days ago. This
storage limitation will be of no use.
DND
Do not disturb, once checked, the extension will be unavailable.
Call Recording
Check this option to record all interior calls made by this extension, user can check the recordings on the personal
web portal and with record account (password is the same with admin) too.
3.2 SIP extensions
SIP extension is a SIP Account that allows an IP Phone or an IP SoftPhone client to register on IPPBX. It can be
created / modified / deleted one by one or in batch.
3.2.1 Add new extensions
a. Add single extension
Click to add an extension
NOTE: SIP extension number range is defined in PBX Settings -> Options -> Extension Preference, changing
it can create extensions in others number range.
1) General
Type
SIP
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.
Extension
Extension number, i.e. 6000, it is associated with this particular user / phone.
Password
Authentication for SIP phone to register and make calls.
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2) SIP Setting
NAT
Try this setting when IPPBX 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, IPPBX will route the media streams from SIP endpoints through itself. Enabling this option causes
IPPBX to attempt to negotiate the endpoints to route the media stream directly, bypassing IPPBX. It is not always
possible for IPPBX 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 IPPBX 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.
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.
SIP Quality
Check this option to force keep alive.
3) 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.
4) 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.
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
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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.
IPPBX 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. IPPBX 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.
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Storage Quota Privilege
Set capacity of disk space for this extension to store voicemail.
Restricted: 1 M
Basic: 2 M
Regular: 3 M
Privileged: 4 M
Super: 5 M
NOTE: If you have set System Settings -> External Storage and set Move Files Created Before 0 days ago.
This storage limitation will be of no use.
DND
Do not disturb, once checked, the extension will be unavailable.
Call Recording
Check this option to record all interior calls made by this extension, user can check the recordings on the personal
web portal and with record account (password is the same with admin) too.
Advanced Configuration
Preferred Codec
Set the allowed codec and priority for SIP phone. The options are below:
Audio: A-law, U-law, GSM, SPEEX, G726, G722, ADPCM, G729
Video: H261, H263, H263P, H264
NOTE: There must be at least one same audio/video codec chose in IPPBX extension settings and SIP phone
codec settings, otherwise, It’s impossible to make audio/video calls between IPPBX and SIP phone.
IP Restriction
Enable it to permit trusted IP/network register to this extension number. This is an useful way to improve the
security of IPPBX.
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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
b. Click to add multiple extensions.
3.2.2 Modify extensions
a. Click behind an extension to edit the extension.
b. Choose one or more extensions, click to modify them simultaneously.
3.3 Register onto IPPBX with your IP phone
Login the web administration page of IP phone and set the account information.
After successfully register with 6000 and 6001, you can make interior calls among 6000, 6001, 6101(FXS) now.
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3.4 Extensions Status
This page is used to check the extensions status.
Idle: The extension is registered and idle.
Busy: The extension is on the phone.
Ringing: The extension is ringing.
Hold: The extension is on hold
Unavailable: The extension is not registered and unreachable.
NOTE: If this page response slowly, please be patient to wait the output before check other pages. Otherwise,
other pages cannot be displayed correctly since IPPBX is accessing database while status checking, and
database is locked for other pages’ request.
3.5 Feature Codes
1) General
Call Recording
Record a call while in the call.
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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.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
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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.
NOTE: Default Follow Me settings are as below:
Always: Disabled
When no answer: Enabled
When busy: Enabled
Forward to: Voice Mail
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
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Set the destination for call forward to voicemail.
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 IPPBX 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
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.
NOTE: After SMTP setting, please set Email address for each extension to achieve Voicemail to Email function.
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
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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 IPPBX 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.
Delete Messages After Notification
When voicemail to email is set, after voicemail is sent out via email successfully, the voicemail will be deleted
from IPPBX. This is intended for use with users who wish to receive their voicemail ONLY by email.
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, IPPBX will playback MoH for the first participant.
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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
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.
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2) Extension Preference.
Defines the range for VOIP / 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:
VOIP extension
100
FXS extension
8
Ring Group
9
Voice menu/IVR
16
Paging Group
9
Conference
9
Queue
9
4. Create SIP trunk to communicate with VoIP provider
In this page, we can also configure the ‘service provider’ trunk, which doesn’ t need the use name and
password for authorization, when you have bought a trunk from provide with IP address only, please choose
‘service provider’ trunk .
4.1 Create SIP trunks
Go to PBX Settings -> Trunks, Click to add a new SIP trunk.
Type
SIP
Trunk Name
A unique label to help you identify the trunk.
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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.
Call Recording
Check this option to record all exterior calls made by this trunk. User can check the recordings with record
account(password is the same with admin) .
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Advance Configuration
DOD(Direct Outward Dialing Number) Setting
Set the Outbound number for different extensions.
For example:
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.
< === 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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4.3 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.
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 6.
Prepend these digits: Allows the user to specify digits that are prepended before the call is placed via the trunk.
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.
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4.4 Make inbound calls
Go to PBX Settings -> Inbound Routes, click to add an inbound route.
1)General
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.
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. The DID for SIP trunk is usually as same as the account (others can only dial the account
number to dial into PBX), then just leave it blank is OK .
Extension
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Define the extension for DID number. This field is usually unavailable for most VoIP trunk, if you VoIP provider
support DID, you can set it. 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.
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.
2)Inbound Trunk Selection
Select the trunks for which this inbound route apply.
3)Time
Select appropriate time intervals for when this inbound route apply.
4)Path
Set the destination for incoming calls. If Extension is set, this option will not take effect.
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5. Create Service Provider to connect two IP PBXes
Service Provider is used to interconnect two IP PBXes, it must be created in both IP PBXes, after setting service
provider and outbound route, extensions of each IP PBX can call the other’s directly according to the outbound
route.
5.1 Create Service Provider
Go to PBX Settings -> Trunks, Click to add a new service provider.
Type
SIP
Provider Hostname/IP
Hostname or IP of your VoIP provider, default port is 5060.
Provider Name
A unique label to help you identify the service provider.
Call Recording
Check this option to record all calls made by this trunk. User can check the recordings with record
account(password is the same with admin) .
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.
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.
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. Detailed example can be found in Chapter 4.1.
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5.2 Check Service Provider Status
After setting service provider in the other PBX, the status will become to Registered.
5.3 Make outbound calls
Go to PBX Settings -> Outbound Routes, click to add an outbound route.
The setting here is as same as that for SIP trunk, please refer to Chapter 4.3 to set it.
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Then 6000/6001 can dial 900-999 to corresponding extension (900-999) in other end directly.
5.4 Make inbound calls
Go to PBX Settings -> Inbound Routes, click to add an inbound route.
Even there is no inbound route set, two IPPBXes can communicate with each other, however we can make it
more functional with setting DID to those differ from local extensions.
Example 1: DID + Extension--Make calls more easier.
DID: 00-03 Extension: 6000-60003
Extensions from other end can dial 6000-6003 extension directly to reach these extensions, besides, they can
dial 00-03 to reach them if their outbound route allow. In this case, Path is of no use.
Example 2: DID + Path--Forward calls to other applications.
DID: 6600 Path: Ring Group
Extensions from other end can dial local extensions directly, besides, they can dial 6600 to reach the destination
set in Path. In this case, they can dial 6600 to reach the ring group.
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6. Make outbound/inbound calls to/from PSTN network
6.1 Make sure FXO modules are installed
If there are FXO modules installed in your IPPBX, IPPBX configures analog trunk automatically when they are
detected.
NOTE: Before using them, please make sure FXO port is connected with PSTN line (InService). The connection
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status can be checked in System Status -> Trunk Status
6.2 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
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.
For example: If set Dial Pattern: 9, Strip: 1, Prepend 123, user need to dial 94567 to dial PSTN number 1234567
Prepend these digits
Allows the user to specify digits that are prepended before the call is placed via the trunk.
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 6000,6001 can prefix 9 to dial out.
Outbound Trunk Selection
Select trunks which calls are forwarded out through. If there are more than one FXO trunk chose, PBX will try to
dial out through next trunk if the previous one failed.
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6.3 Make inbound calls
Go to PBX Settings -> Inbound Routes, click to add an inbound route. Just setting Selected Trunks
and Path is OK.
1) General
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
It’s unavailable for Analog trunk, leave it blank.
2) Inbound Trunk Selection
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Select the trunks for which this inbound route apply.
3) Time
Select appropriate time intervals for when this inbound route apply.
4) Path
Set the destination for incoming calls.
7. Inbound Call Control
7.1 Time Interval
Set the Time Interval for inbound route.
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7.2 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.
Ignore Call Forward
By default, if there is call forward enabled in IP phone, IPPBX will call that forwarding number. when this option is
checked, IPPBX will ignore that call forward setting.
Enable MoH
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When MoH is choose, caller will hear MoH other than ring tone.
Parallel
Ring all the members at the same time.
Serial
Ring all the members one by one.
No answer forward to
Set the failover destination if there is no answer.
7.3 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 IPPBX 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
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
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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
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.
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.
Retry
How long does IPPBX 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.
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7.4 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.
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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 IPPBX to wait user’s input
Repeat Count
How many times to play prompt
Allow Dialing of Other Extensions
Allow dialing local extensions
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Key Press Event
Dial digit to trigger corresponding event
No Entry Forward to
The destination for incoming call if there is none input
Invalid Forward to
The destination for incoming call if there is invalid input
7.5 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 IPPBX 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 IPPBX, IPPBX-A and IPPBX-B, they are connected with SIP trunk, and
IPPBX-A has FXO trunk to connect local PSTN and outbound route for that, DISA can be used as below:
1. Create a DISA in IPPBX-A including the FXO trunk.
2. Set it as the destination of inbound route for SIP trunk.
After users of IPPBX-B dial into DISA application in IPPBX-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.
7.6 Call Back
A callback will hang up on the caller and then call them back, directing them to the selected destination. This is
useful for reducing mobile phone charges as well as other applications. Outbound calls will proceed according to
the dial patterns in Outbound Routes
Callback Name
A name for the Callback
Callback Number
Enter the number to dial for the callback. Leave this blank to just dial the incoming Caller ID Number.
Delay before Callback
Enter the number of seconds the system should wait before calling back.
Call Recording
Check this option to record all calls made by this Call back. User can check the recording with record
account( password is the same with admin).
PATH
Choose the destination which IPPBX will bridge to caller.
For Example:
1. Set Callback:
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2. Set inbound route for FXO trunk:
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3. Dial into IPPBX from PSTN number, and hang up immediately once the call is connected. Then IPPBX dial
extension 6002, after 6002 answer the call, IPPBX dial caller and bridge them.
7.7 Inbound to Outbound
Outbound Routes can act as destination in Inbound route settings, it let two ends connecting to IPPBX with trunk
communicate directly.
7.7.1 FXO trunk to SIP trunk
1. Set outbound route for SIP trunk.
2. Set Inbound route for FXO trunk.
Destination
Choose the proper SIP trunk
Number
Set a number on the other end of SIP trunk. it must be set.
3. Dial into IPPBX from PSTN, then IPPBX will dial 202 through SIP trunk.
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7.7.2 SIP trunk to FXO trunk
1. Set outbound route for FXO trunk.
2. Set Inbound route for SIP trunk/Service Provider. Since service provider is more flexible, here we use Service
Provider as an example
Destination
Choose the proper FXO trunk
Number
Set a number on the other end of FXO trunk. it can be blank, then it means the number caller dialed.
3. Dial into IPPBX from SIP trunk, e.g. 83018618, then IPPBX will dial that number through FXO trunk selected
in PSTN outbound route.
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7.8 Custom Dialplan
Click to upload Custom Dialplan. There are three types file: user, dialplan and
dahdi.
user
This file aimed at defining a "user". It can define a user with an optional SIP phone, Dahdi phone and/or just
about any other type of phone.
dialplan
This file contains the "dial plan" of Asterisk, the master plan of control or execution flow for all of its
operations. It controls how incoming and outgoing calls are handled and routed. This is where you configure
the behavior of all connections through IPPBX.
dahdi
Asterisk cmd DAHDILookup configuration.
NOTE: You must prefix custom_ to your custom file and context to avoid conflict with the system
file,users.conf,extensions.conf and chan_dahdi.conf.
7.9 Blacklist
Block incoming calls from specified numbers
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If a number in blacklist dial into IPPBX, 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.
7.10 SIP Settings
1) General
UDP Port
Set the SIP port (UDP) which IPPBX is listening to.
Enable TCP
Enable TCP protocol for SIP.
TCP Port
Set the SIP port (TCP) which IPPBX 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.
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
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b. NAT
The External IP, External Host and Local Network Identification settings are used if you use IPPBX 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 IPPBX will perform DNS queries periodically. Not
recommended for production environments! Use External IP instead.
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 IPPBX to stay in the
media path, the media will be redirected. This does not really work with in the case where IPPBX is outside and
have clients on the inside of a NAT. In that case, you want to set this option to no nat.
8. Audios
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8.1 Music On Hold
Manage audio files for Music On Hold, the format should be .WAV (16 bit, mono 8000 Hz) and .GSM, the size
should less than 4 MB.
8.2 Custom Prompts
Manage prompts used for Voice Menu. It can be recorded by extensions or uploaded from local PC, the format
should be .WAV (16 bit, mono 8000 Hz) and .GSM, the size should less than 4 MB
8.3 Language Setting
Set the language of default system prompt audio, English is supported by default. French, Italian, Russian and
Spanish need to be download from Internet when chose at the first time. Make sure gateway is right set so that
IPPBX can access Internet.
9. Network Settings
Description of WAN settings can be found in Chapter 2. All network settings will take effect immediately when you
save the change.
9.1 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.
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9.2 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
Set the destination port range for connection. The main protocols and default ports IPPBX uses for each
application are list below:
HTTP
TCP:80
HTTPS
TCP:443
SIP
UDP:5060
SIP
TCP:5060
RTP
UDP:10000-20000
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.
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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
4) SIP Defense:
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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
Check this to drop all packets or connection from other hosts if there are no other rules defined.
6) Firewall Setting Example :
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9.3 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 4 DDNS servers below, please go to the website of below servers and apply a
domain name then fill related information here.
dyndns.org
dyndns.com
qdns
www.zoneedit.com
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9.4 VLAN
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.
9.5 VPN
A virtual private network (VPN) extends a private network across a public network, such as the Internet. It enables
a computer to send and receive data across shared or public networks as if it were directly connected to the
private network, while benefitting from the functionality, security and management policies of the private network.
This is done by establishing a virtual point-to-point connection through the use of dedicated connections,
encryption, or a combination of the two. Currently, only PPTP is supported.
10. System Settings
10.1 Change Password
Change the password for admin login, it will take effect immediately.
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10.2 Auto Provisioning
IP01 can only configure ATCOM AT8XX serials IP phone via SIP Subscribe + Notify way. Make sure your AT8XX
IP phone is using the latest firmware.
10.2.1 Auto Provisioning
1) Phonebook
You can add your contacts here and when you use phone provisioning, IP phone will download the phone
book.
Add
Name: Input the name of this contact.
Type: These are two types: Directory and Blacklist.
Number: Input the number here.
Upload phonebook
You can upload a phonebook before auto provision, which will be provisioned to the IP phone when using
auto provision feature to configure your IP phones. The format of phonebook should be *.xml.
NOTE: All the existing phonebooks of the IP phone will be replaced automatically if the phonebooks are
configured in this way.
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2) Ring Tone
You can upload a Ring Tone before auto provision, which will be provisioned to the IP phone when using
auto provision feature to configure your IP phones.
NOTE: Ring tone must be wav file, 8k sampling rate, 8 Bit u-law compression. File size should < 200Kbytes.
3) Firmware
You can upload a firmware before auto provision, which will be provisioned to the IP phone when using auto
provision feature to configure your IP phones.
NOTE: the file format must be .fw
10.2.2 Configured Phones
Create New Phone
Enable
Enable auto provisioning for this phone
Phone Type
Choose the Phone Type
MAC Address
Input the MAC address for the phone
New Version
Make sure your phone is with the newest version. If yes, IPPBX will generate the configuration file for IP phone.
Manufacturer
Choose the manufacturer.
DND
Enable or Disable DND.
Call Waiting
Enable or Disable Call Waiting.
Auto Answer
Enable or Disable Auto Answer.
Phone book
Enable or Disable Phone book update .If this option is checked, you must add or upload phone book first.
Ring Tone
Enable or Disable Ring Tone. If this option is checked, you must upload phone book first.
Firmware
Enable or Disable Firmware update. If this option is checked, you must upload firmware first.
Firmware version
If Firmware option is enable, you must input the correct firmware version here.
NOTE: If the firmware version is different from the one you’re going to load in IP phone, it will cause
problems in IP phone.
Line
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You can configure the line key settings for this IP phone.
10.2.3 Unconfigured Phones
In this section, IPPBX will scan all the supported IP phones and display here, you can click the ‘Edit’ button
of IP phone and input the corresponding information in the pop-up window.
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10.2.4 Upload Phone Configure
Click “Upload a file” and choose the configure file of IP phone in the popup window.
NOTE: the file format must be .xml.
10.3 Date && Time
Set the date and time for IPPBX. The settings will take effect immediately.
1) General
There are two ways to set Date/Time for IPPBX:
a. NTP server
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Make sure the connection between IPPBX and NTP server is OK, if the NTP server is located on Internet, the
gateway of WAN should be right set so that IPPBX 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 IPPBX time from the first Sunday coming after April 1th to
the first Sunday coming after October 31th will be one hour early.
10.4 External Storage
The External Storage feature is used to extend storage space. Once configured, the files (voicemail/Call
recording (>60M) / CDR(>5M) ) created before the configured days will be moved to the Net-Disk.
NOTE: The shared folder must be based on Windows operation system.
1. Choose a window-based computer that is always in service
2. Create a folder
3. Share this folder
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4. Set External Storage
Net-Disk Host/IP
IP of the PC
Net-Disk Share Name
The name of the share folder
Net-Disk Access User Name
Account in that PC
Net-Disk Access Password
Password for the account, if there is no password required, leave it blank.
Backup Period
How often PBX move its voicemail/call recording/CDR data to PC
Move Files Created Before days ago
Choose what files should be move to PC according to their creation time. 0 means immediately. Note that even 0
is set, the size of call recording must be more than 60 M, that of CDR must be 10M. Then IPPBX will move them
to PC.
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10.5 Firmware Upgrade
Firmware for IPPBX There are two ways to upload Kernel / Application for IPPBX:
1) HTTP
Upload them from local PC. It will reset IPPBX to factory default settings by default.
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.
10.6 Backup and Restore
1) Backup
Create Backup for configuration / System audio prompt / Voice Mail. The backup can be downloaded to local PC.
IPPBX IP01 User Manual
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2) Restore
Click Restore to restore corresponding backup, backup file can be uploaded from local PC. Backup will be used
after IPPBX reboot. It can’t be used for different product models.
10.7 Reboot && Reset
Reboot or Reset IPPBX.
You need to choose the product model and extension length/prefix the first time to login IPPBX after it is
reset to factory default settings.
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11. Reports
11.1 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
IPPBX supports two delete operations: delete selected CDR and delete all CDR.
3) Download
It can be downloaded to local PC
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11.2 Syslog
1) Syslog
Set the Syslog level and download it.
2) Packet Capture Tool
This feature always used to capture packets for technician. Users could specify the destination IP address
and port to get the packets.
IP
Specify the destination IP address to get the packets.
Port
Specify the destination port to get the packets.
12. Web Interface for extension
PBX allows users to check their voicemail /Call Recording/CDR, and set personal settings.
IPPBX IP01 User Manual
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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.
2) Call Recording Checking
IPPBX IP01 User Manual
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3) CDR Checking
Users can check their CDR here.
4) Personal Settings
Users can set voice mail / voice mail to email / follow me / ring timeout here.
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