FIBERME Communications LLC.

FCM5404 IP PBX – User Manual

 Table of Content

Table of Content .................................................................................... i

Copyright Declaration ......................................................................... iv
Revision History ................................................................................... v
Chapter 1 Product Introduction ...................................................... 1
1.1. Typical Application ......................................................................................... 1
1.2. Feature List .................................................................................................... 2

Chapter 2 Quick Guide .................................................................... 5

Chapter 3 WEB Configuration ........................................................ 6
3.1. System Login ................................................................................................. 6
3.2. System Status................................................................................................ 6
3.2.1. Dashboard .................................................................................................................. 6
3.2.2. System Info ................................................................................................................ 8
3.2.2.1. General Info ........................................................................................................ 6
3.2.2.2. Network ............................................................................................................... 6
3.2.2.2.1. LAN ................................................................................................................. 6
3.2.2.2.2. WAN ..................................................................................................................... 7
3.2.2.3. Performance ....................................................................................................... 7
3.2.2.4. Storage Usage .................................................................................................... 7
3.2.3. PBX Monitor ............................................................................................................... 8
3.2.3.1. Extension ............................................................................................................ 8
3.2.3.2. Trunk......................................................................................................................... 8
3.2.4. Active Call Queue ...................................................................................................... 8
3.2.5. Active Calls................................................................................................................. 9
3.2.6. Active Conferences .................................................................................................... 9
3.2.6.1. Conference List ................................................................................................... 9
3.2.6.2. Conference Contacts .......................................................................................... 9
3.2.7. Multicast Monitor ........................................................................................................ 9
3.2.8. View Voicemail ........................................................................................................... 9
3.3. Extensions/Trunks ....................................................................................... 10
3.3.1. Extensions .................................................................................................................10
3.3.1.1. Extensions......................................................................................................... 10
3.3.1.1.1. Basic ............................................................................................................. 10
3.3.1.1.2. Features ........................................................................................................ 10
3.3.1.1.3. Advanced ...................................................................................................... 11
3.3.2. Extension Groups ..................................................................................................... 14
3.3.3. Trunks ............................................................................................................................ 14
3.3.3.1. Basic ................................................................................................................. 14
3.3.3.2. CODEC ............................................................................................................. 15
3.3.3.3. Advance ............................................................................................................ 15
3.3.3.4. DOD .................................................................................................................. 17
3.3.3.5. Adapt Caller ID .................................................................................................. 17
3.4. Call Control ................................................................................................... 18
3.4.1. Inbound Routes ........................................................................................................ 18

Page i
 3.4.2. Outbound Routes ..................................................................................................... 21
3.4.3. Outbound Restrictions.............................................................................................. 23
3.4.4. AutoCLIP .................................................................................................................. 23
3.4.5. CC Routes ................................................................................................................ 24
3.4.6. Time Condition ......................................................................................................... 24
3.4.7. FXO Lines Locked.................................................................................................... 24
3.5. Call Features ............................................................................................... 25
3.5.1. IVR ........................................................................................................ 25
3.5.1.1. IVR .................................................................................................................... 25
3.5.1.2. Conference........................................................................................................ 26
3.5.1.2.1. Basic ............................................................................................................. 26
3.5.1.2.2. Advanced Settings ........................................................................................ 26
3.5.1.2.3. Appointment Meeting Settings ..................................................................... 26
3.5.1.3. Call Queue ........................................................................................................ 27
3.5.1.3.1. Basic ............................................................................................................. 27
3.5.1.3.2. Caller Experience Settings ........................................................................... 28
3.5.1.3.3. Key Press Event ........................................................................................... 29
3.5.1.4. Pickup Group .................................................................................................... 29
3.5.1.5. Ring Group ........................................................................................................ 29
3.5.1.6. Blocked List ....................................................................................................... 30
3.5.1.7. PIN Numbers..................................................................................................... 31
3.5.1.8. Speed Dial......................................................................................................... 31
3.5.1.9. Paging ............................................................................................................... 31
3.5.1.10. DISA .................................................................................................................. 31
3.5.1.11. Call Back ........................................................................................................... 32
3.5.1.12. Emergencies ..................................................................................................... 32
3.5.1.13. Multicast ............................................................................................................ 33
3.5.1.14. Wakeup Services .............................................................................................. 33
3.6. Billing ........................................................................................................... 42
3.6.1.1. Extensions......................................................................................................... 42
3.6.1.2. Rates ................................................................................................................. 42
3.6.1.3. Top-up History ................................................................................................... 42
3.6.1.4. Statistics ............................................................................................................ 42
3.6.1.5. Billing Settings .................................................................................................. 43
3.7. FCM Setting ................................................................................................. 43
3.7.1. SIP Settings .............................................................................................................. 39
3.7.2. Record Settings ........................................................................................................ 38
3.7.3. Voice Prompts .......................................................................................................... 36
3.7.3.1. Voice Prompts ................................................................................................... 36
3.7.3.2. System Prompt ................................................................................................. 36
3.7.3.3. Music on Hold ................................................................................................... 37
3.7.3.4. Custom Prompt ................................................................................................. 37
3.7.4. Storage ..................................................................................................................... 46
3.7.4.1. Preference......................................................................................................... 46
3.7.4.1.1. Storage Locations ......................................................................................... 46
3.7.4.1.2. Storage Devices ........................................................................................... 46
3.7.4.2. Auto Cleanup .................................................................................................... 46
3.7.4.2.1. CDR Auto Cleanup ....................................................................................... 46
3.7.4.2.2. Voicemail and One Touch Recording Auto Cleanup ..................................... 47
3.7.4.2.3. Recordings Auto Cleanup ............................................................................. 47
3.7.4.2.4. Logs Auto Cleanup ....................................................................................... 47
3.7.4.3. Backups ............................................................................................................ 47
3.7.5. Preference ................................................................................................................ 38
3.7.6. Voicemail .................................................................................................................. 37
3.8. System Setting............................................................................................. 38

Page ii
 3.8.1. Security Strategy ...................................................................................................... 44
3.8.1.1. Security Strategy............................................................................................... 44
3.8.2. Network Settings ...................................................................................................... 43
3.8.2.1. Basic Settings ................................................................................................... 43
3.8.2.2. OPENVPN......................................................................................................... 44
3.8.2.3. Static Routes ..................................................................................................... 44
3.8.3. LDAP Server ............................................................................................................ 45
3.8.4. Phone Book .............................................................................................................. 43
3.8.5. Email Settings .......................................................................................................... 49
3.8.6. Service ..................................................................................................................... 45
3.8.7. Event Setting ............................................................................................................ 48
3.8.7.1. System Settings ................................................................................................ 48
3.8.7.2. PBX Settings ..................................................................................................... 49
3.8.7.3. Notification Contacts ......................................................................................... 49
3.8.8. Feature Code ........................................................................................................... 33
3.8.9. Date Time Settings ................................................................................................... 45
3.9. Maintenance ................................................................................................ 50
3.9.1. Upgrade.................................................................................................................... 50
3.9.2. Users ........................................................................................................................ 48
3.9.3. User Group ............................................................................................................... 48
3.9.4. Reboot ...................................................................................................................... 50
3.9.5. Backup and Restore................................................................................................. 50
3.9.5.1. Backup .............................................................................................................. 50
3.9.5.2. Restore.............................................................................................................. 50
3.9.5.3. Backup Lists ...................................................................................................... 51
3.9.6. Factory Reset ........................................................................................................... 51
3.9.7. Operation Log........................................................................................................... 51
3.9.8. PBX LOG.................................................................................................................. 51
3.9.9. Log Viewer ............................................................................................................... 51
3.9.10. Trouble Shooting ...................................................................................................... 51
3.9.11. Event Log ................................................................................................................. 52
3.10. Call Detail Records ...................................................................................... 10
3.10.1. Call Detail Records................................................................................................... 10
3.10.2. Extension Summary ................................................................................................. 10
3.10.3. Conference Recording Session ............................................................................... 10
3.11. Value Added Features ................................................................................. 50
3.11.1. Authorization ............................................................................................................ 52
3.11.1.1. Authorization Info .............................................................................................. 52
3.11.1.2. Upload Authorization File .................................................................................. 52
3.11.1.3. Clear Authorization............................................................................................ 52
3.11.2. Auto Provision .......................................................................................................... 41

Page iii
 Copyright Declaration

All rights reserved; no part of this document may be reproduced or transmitted in any form or by
any means, electronic or mechanical, without prior written permission from FIBERME
Communications LLC (hereinafter referred to as ‘FIBERME’).

FIBERME reserves all rights to modify this document without prior notice. Please contact
FIBERME for the latest version of this document before placing an order.

FIBERME has made every effort to ensure the accuracy of this document but does not guarantee
the absence of errors. Moreover, FIBERME assumes no responsibility in obtaining permission
and authorization of any third-party patent, copyright or product involved in relation to the use of
this document.

Page iv
Chapter 1 Product Introduction
Thank you for choosing the FIBERME IPPBX Series products which provide excellent VoIP
solutions for Enterprise Unified Communications, Customer Service Center, Hotel Voice
Communications, etc.

1.1. Typical Application

Figure 1-1IPPBX Typical Application

Main functions in typical applications:

Enterprise Unified Communications: Extension, trunk, route, CDR, call recording, IVR,
voicemail, teleconference, call forwarding, time condition, monitoring, mobile extension,
intercepting, etc.
Customer Service/Call Center: Extension, trunk, route, CDR, call recording, queue, monitoring,
call forwarding, time condition, etc.
Hotel Communications: Extension, trunk, switchboard, wakeup service, paging, etc.

Page 1
1.2. Feature List
Basic Features Description

Allow users to make calls from extension to extension after registering SIP
Extension
extensions to IPPBX.

Allow extension users to make incoming and outgoing calls by SIP and FXO
Trunk
trunks with the help of inbound and outbound routes.

Enable forwarding calls from SIP or FXO trunks to internal extensions, IVR,
Inbound Routes
conference, call center, DISA, callback systems, etc.

Outbound Routes Enable making calls from extensions to external PSTN users.

Allow users to query and download detailed call records by condition on the
CDR
webpage.

Record extensions, trunks, conferences, call centers; query, play and

Call Recording
download the recording.

Extensions can be forwarded on different conditions such as ‘Always’, ‘On

Call Forwarding Busy’, ‘No Answer’, or ‘Not Registered’. Meanwhile, time condition settings
are supported.

This feature allows an FXS extension to receive another call while on the
Call Waiting
phone. It will make the feature of transfer on busy invalid.

If an extension on the FXS port doesn’t dial out within the set time after it is
Hotline
picked up, the preset number will be called automatically.

Do Not Disturb Reject all incoming calls to this extension.

Multiple mobile numbers can be set for an extension to avoid missing any call
Mobile Number
to it.

Support monitoring modes All, Listen, Whisper, Barge-in and monitoring

Monitor
authorities Disable, Enable All, Extensions to set for an extension.

Each extension supports an independent voicemail box as well as sending

Voicemail
messages to a designated E-mail address.

Fax Support T.38 fax extension and fax gateway modes.

Extension Security Guarantee the security of extensions by password, ACL, User Agent, etc.

Enable a connection of the station which is linked with the FXS port and the
Communication
trunk which is linked with the FXO port to keep the calls between the FXS and
without Power
FXO ports uninterrupted during power outage.

IVR Customize multi-level IVR.

Customize call center queues, providing multiple station ringing strategies to

Call Center Queue
satisfy a variety of applications.

Conference Support teleconferencing with more than 30 parties.

AutoCLIP Redirect call to original extension.

When the extension is busy, the call will be recorded. After the callback
CC Routes
interval, the call will be dialed back.

Page 2
 Set a group of extensions into a ring group. When the callers call the ring
Ring Groups group, all available extensions will ring simultaneously or sequentially (up to
different ringing strategies).

Support interception of inside calls in a group and calls of specified

Intercept Groups
extensions.

Call Paging Meet such requirements as paging system.

Allow users to “park” a phone call with a parking extension number, placing it
Call Parking on hold to be answered on a softphone or any other phone in the office. The
caller is put on hold while users switch phones.

Numbers in the blacklist will be blocked to call in, or called, or both. It supports
Blacklist
two modes: Exact Match and Regex Match.

Enable outside users using PBX service just like the system extensions to
DISA
make calls.

Callback Hang up the specified callers and let the PBX call them back.

Customize a short number that allows fast dialing of your frequently used
Speed Dial numbers so that you can place a call by pressing a reduced number of keys
without having to look up his/her phone number.

Emergency Number You can call the emergency number through the designated relay.

You can customize the time and the extension, and then the PBX will initiate a
Wakeup Service
call to the extension

You can customize the time and the multicast address, and then the PBX will
Multicast
play a prompt to the corresponding multicast address.

This feature is supported for inbound routes, call forwarding, mobile

Time Condition
extensions, etc.

This feature is supported for outbound routes, DISA, conference, voicemail,

PIN Code
etc.

Signaling & Protocol Description

SIP Signaling Supported protocol: SIP V1.0/2.0, RFC3261

CODEC G.711A, G.711U, G.729, G722, G723, G726-16,
speex@8000h@20i, speex@16000h@20i, AMR,
Voice
PCMA@30i, PCMA@40i, PCMU@30i, PCMU@40i
DTMF Mode RFC2833, RFC4733, SIP INFO, INBAND

Network Description

Supported protocol: TCP/UDP, TLS, SSH, HTTPS, ARP/RARP, DNS, NTP,

Network Protocol
TFTP, TELNET, STUN.

Static IP IP address modification support.

DHCP IP address dynamic allocation support.

DNS Domain Name Service support.

Security Description

ACL This feature is supported for extension registration and WEB access, etc.

Page 3
 Allow users to customize dynamic firewall strategies to guarantee the security
Auto Defense
of system and network.

TLS&SRTP Guarantee the security of signaling and voice communications.

Maintain & Upgrade Description

WEB Configuration Support of configurations through the WEB user interface.

Language Chinese, English.

Support of user interface, IPPBX service, kernel and firmware upgrades based
Software Upgrade
on WEB.

Tracking Test Support of Ping and Tracert tests based on WEB.

SysLog Type ERROR, WARNING, NOTICE, INFO, DEBUG, CONSOLE

Page 4
 Chapter 2 Quick Guide
This chapter is intended to help you grasp the basic operations of the FCM IPPBX product in the
shortest time.
Step 1: Confirm that your packing box contains all the following things.
⚫ FCM5404 *1
⚫ FCM5404: External 12V Power Adapter *1
⚫ Warranty Card *1
⚫ Installation Manual *1
Step 2: Connect the network cable.
Connect the LAN port of FCM5404 with the network cable of the PC, or connect it to the router or
PBX. Configure the IP address of the PC to 192.168.0.200 and then you can go
https://192.168.0.101 to visit the webpage of FCM5404.
Go to the page Network Settings to configure the actual IP address, subnet mask, gateway, etc.
Then use the modified IP to visit the webpage of FCM5404.
Step 3: Add and configure SIP extensions.
Go to the page Extensions to add SIP extensions. Modify extension settings and enable
necessary functions according to your requirements. After that, you can perform a dial from
extension to extension.
Step 4: Add and configure SIP trunks.
Go to the page Trunks to add SIP trunks and modify trunk settings according to your
requirements.
Step 5: Add call features.
Go to the page Call Features to add necessary call features, such as IVR menus, conference
rooms, call center queues, ringing groups, etc.
Step 6: Add inbound routes.
Go to the page Inbound Routes to add inbound routes and set route destinations, such as
extensions, IVR menus, conference rooms, call center queues, ringing groups, etc.
Step 7: Add outbound routes.
Go to the page Outbound Routes to add outbound routes and set member extensions for each
route.

Special Instructions:
⚫ The chassis of the FCM5404 IPPBX product must be grounded for safety reasons,
according to standard industry requirements. A simple way is earthing with the third pin
on the plug or the grounding studs on the machine. No or improper grounding may cause
instability in operation as well as decrease in lightning resistance.
⚫ As the device will gradually heat up while being used, please maintain good ventilation to
prevent sudden failure, ensuring that the ventilation holes are never jammed.
⚫ During runtime, if the SYS indicator doesn’t flash regularly and you cannot figure out
and solve the problem by yourself, please contact our technicians for help. Otherwise, it
may lead to a drop in performance or unexpected errors.

Page 5
 Chapter 3 WEB Configuration

3.1. System Login

Make sure the LANs of PC and IPPBX are in the same network segment. Enter the default IP
address of IPPBX https://192.168.0.101 to log in the web interface.

The original username and password are both admin. After login, you can add users and set
users’ access authority, as well as modify the username and password.

Note: We suggest you use those browsers Chrome 67, Firefox60, IE11 or above versions to
ensure the normal access of the management interface.

WANIP: 192.168.1.101;

LANIP: 192.168.0.101.

3.2. System Status

It includes eight parts: Dashboard, System Info, PBX Monitor, Active Call Queue, Active Calls,
Active Conferences, Multicast Monitor and View Voicemail.

3.2.1. Dashboard
The FCM5404 monitors the status for Trunks, Interfaces, Storage Device Life. It presents
administrators the real-time status in different sections

3.2.2. System Info

3.2.2.1. General Info

Item Description
System Time Current system time of IPPBX
Up Time Running time of IPPBX since startup
Product FCM5404
Serial Number Unique identifier of the device
Max Sessions The default value for FCM5404 is 30. It can be authorized.

Max Extensions The default value for FCM5404 is 150. It can be authorized.

uboot Version information of the current uboot

kernel Version information of the kernel
version Version information of the current software

Page 6
3.2.2.2. Network
3.2.2.2.1. LAN
Item Description
TYPE Static IP
MAC MAC address of LAN
IP Address IP address of LAN
Gateway Gateway address which displays only when LAN is the default network interface
Subnet Mask Information about subnet mask
Preferred DNS Server Information about preferred DNS server
Alternate DNS Server Information about alternate DNS server
When the network cable is well connected and the network goes normal, here
Alternate DNS Server
displays connection. If the network cable is not connected or the network is
unreachable, here displays disconnection.

3.2.2.2.2. WAN
Item Description
TYPE Static IP, DHCP or PPPoE
MAC MAC address of WAN
IP Address IP address of WAN
Gateway Gateway address which displays only when WAN is the default network interface
Subnet Mask Information about subnet mask
Preferred DNS
Information about preferred DNS server
Server
Alternate DNS
Information about alternate DNS server
Server
When the network cable is well connected and the network goes normal, here
Network State displays connection. If the network cable is not connected or the network is
unreachable, here displays disconnection.

3.2.2.3 Performance
Item Description
CPU Real-time display of current CPU utilization
MEMORY Real-time display of current memory utilization
LAN Real-time display of current rate of LAN
WAN Real-time display of current rate of WAN

3.2.2.4. Storage Usage

Item Description
Flash Display of total and used storage of the built-in flash card as well as the utilization
TF Display of total and used storage of the outer TF card as well as the utilization
USB Display of total and used storage of the outer USB card as well as the utilization
NetDisk Display of total and used storage of the network disk space as well as the utilization

Page 7
3.2.3. PBX Monitor

3.2.3.1. Extension
Item Description
For a SIP trunk, display of status: unregistered/registered/ringback/ringing/talking;
Status
for an FXO trunk, display of status: idle/ringback/ringing/talking.
Extension Extension number
Name Name of the extension user
Type Extension type, FXS or SIP
For a SIP trunk, display of IP address and port number; for an FXS trunk, display of
IP and Port
physical port number.
3.2.3.2. Trunk
Item Description
Trunk Name User-defined name of the trunk
Type Trunk type, FXO or SIP
For an FXO trunk, display of status: unusable/idle/in use; for a SIP peer trunk,
Trunk Status display of status: unmonitored/unusable/usable; for a SIP Register trunk, display of
status: fail to register/registered.
For a SIP extension, display of domain name/IP address of the registered IP/Soft
Domainname/IP/Port
phone; for an FXO extension, display of physical port number.

3.2.4. Active Call Queue

Item Description
Statistics for queue connection rate, total number of calls, number of connected
Active Call Center calls, number of waiting calls, number of abandoned calls, average waiting time,
average talking time.
Click to view the statistics information about the agents in the queue (Extension,
Agents Total Calls, Answered, Missed, Caller Hangup While Agent Ringing, Login Time,
Talk Time, Agent Type, Agent Status)
Click to view the call status information (Status, Caller, Called, Location, Wait Time,
Queue
Talk Time), and you can hang up, monitor and transfer the current call.

Page 8
3.2.5. Active Calls
Item Description
Eavesdrop Click to monitor the current call
Kill Click to end the current call compulsively

3.2.6. Active Conferences

3.2.6.1. Conference List
Item Description
Conference Room
Display the name of the conference room.
Name
Conference Room
Show the number of the conference room.
Number
Show the number of people online in the current conference room. It is empty when
Member Count
there is no conference in progress.
Conference Run Show the duration of the current conference. It is empty when there is no
Time conference in progress.
Click the operation icon of the current conference to display the participating
Operation members in the current conference room, and to further ban, mute, invite, kick out
or add members, etc.
3.2.6.2. Conference Contacts
Item Description
Group Name The name of the conference contacts on the phone book.
Type Two types are available: Extension and Custom.
You can select the extension to add. When the type is Extension, the extension will
Extension
be displayed.
The name of the conference contact. This item is displayed when the type is set to
CID Name
Custom.
The number of the conference contact. This item is displayed when the type is set
CID Number
to Custom.

3.2.7. Multicast Monitor

Item Description
Multicast Name Display the multicast name.
Multicast Number Display the multicast number.
Multicast Type Display multicast types, including Triggle Multicast, Timed Recast and Live
Multicast.
Multicast Address Display the multicast address.
Enabled Whether to enable the multicast feature,
Status Display the state of multicast monitor.

Page 9
3.2.8. View Voicemail
Click to view the voicemail configuration of all extensions. Then click ‘Messages’ following each
extension to view the exact voicemail messages.

3.3. Extensions/Trunks

3.3.1. Extensions
3.3.1.1. Extensions
3.3.1.1.1. Basic
General Description
Type Extension type, SIP or FXS
Extension number consists of all digits, with the default value range of 1000~5899
Extension
which can be modified in ‘PBX->Preference->Extension Preferences’.
It is generated randomly during the creation of a SIP extension and can be modified
Password
by users.
Enabled Set whether to enable the extension or not. By default it is set to true.
Max Registrations Maximum amount of registrations of this SIP extension, with the default value of 3.
Effective Caller ID
The callerID number for this extension to call outbound, i.e. the UserName field.
Number
UserInfo Description
Name The callerID number for this extension to call outbound, i.e. the DisplayName field.
The password for this extension user to log into the system. Username is Name,
User Password
while the default password is ‘Pass’ plus the extension number.
Voicemail Mail To The email address to send voicemail to
Mobile Number Fill in the mobile phone number of this extension user.
The language of voice prompts. Three options available: System Default, Chinese
Prompt Language and English. System Default means to use the same language as set in Voice
Prompts.
3.3.1.1.2. Features
Voicemail Description
Once this feature is enabled, the call to this extension will enter the voicemail if
Voicemail Enabled
failed. By default, the setting is True.
The password to enter the extension voicemail which is a randomly generated value
Voicemail Password
by default and can be modified by users.
Voicemail Keep Set whether to save the voicemail at IPPBX after it is sent with a specified email. By
Local default, the setting is True.
Set the way to send the voicemail. Audio File Attachment: Send the voice message
Voicemail File via email attachment; Download Link: Send the voice message via link. The latter
is the default setting.

Monitor Description
Allow being Set if this extension can be monitored or not.

Page 10
 monitored *Disable: Not allow to be monitored, as default.
*Enable All: Allow all extensions to monitor.
*Extensions: Select extensions to monitor.

Set the mode in which this extension monitors other ones. The default setting is
None
None: You will not be allowed to monitor calls;
All: All the following 3 modes will be available for use;
Monitor Mode
Listen: You can only listen into the call, but cannot talk (default feature code: *54)
Whisper: You can talk to the extension you are monitoring without being heard by
the other parties (default feature code: *55)
Barge-in: You can talk to both parties (default feature code: *56)
Call Forwarding Description
Always redirect calls to the designated destination within the period set by the
Always
following time condition select box. The default setting is Disabled.
Redirect calls to the designated destination if the extension is busy within the period
On Busy
set by the following time condition select box. The default setting is Disabled.
Redirect calls according to call source to the designated destination if not answered
No Answer within the period set by the following time condition select box. The default setting is
Disabled.
Redirect calls to the designated destination if the extension is not registered within
Not Registered the period set by the following time condition select box. The default setting is
Disabled.
Follow Me Description
Bind a target number (internal extension or external number) to this extension.
When there is an incoming call, both original and bind numbers will ring at the same
Follow Me
time so that the agent could pick up the call in different locations. The external
number will go out through SIP trunks.

Do Not Disturb Description

When DND is enabled for an extension, it will reject all incoming calls. The default
Do Not Disturb
setting is Disabled.
Mobility Extension Description
If you enable this setting, when the user’s mobile number dials into the system, the
Enable Mobility phone will have the same user permission as the desktop extension. So the mobile
Extension number will be able to connect with the other extension, dial out with the trunk and
play voicemail.
It is the same with the user’s mobile number. A prefix matching the outbound route
Prefix
also needs to be filled in.
Ring When the extension has an incoming call, it rings on the mobile number
Simultaneously simultaneously and the ‘Follow Me’ function will be invalid.
3.3.1.1.3. Advanced
RTP Settings Description
When this feature is enabled, the RTP stream is encrypted, sharing the same
Enable SRTP
certification with TLS. The default setting is False.

Page 11
 Set whether to send the media stream point to point or in transparent proxy mode.
Proxy Media: The media stream will pass IPPBX (default);
SIP Bypass Media
Bypass Media: The media stream will be transported point to point.
Do not enable recording when set to ‘Bypass Media’ to avoid problems.
Set RTP Codecs. So far G711A, G711U, G729, G722, PCMU, PCMA, GSM, G723,
G726-16, speex@8000h@20i, speex@16000h@20i, AMR, PCMA@30i,
RTP Codec String
PCMA@40i, PCMU@30i, PCMU@40i, VP8, VP9, H264, H263, H263-1998,
H263-2000are supported.
Register Settings Description
Once enabled, only the IP address or IP segment that matches the setting will be
able to register this extension number. For example, 192.168.1.235/24 means all IP
AuthACL
addresses in the segment of 192.168.1 are allowed to register; 192.168.1.235/32
means only the address 192.168.1.235 is allowed to register. By default, it is null.
Send the OPTIONS message to this extension to check if it is registered and
Online Detection
reachable. The default setting is False.
Calculated by second. The default value 0 means using the registration validity of
SIP Force Expires SIP extensions while other values mean compulsively using the registration validity
of IPPBX. Range: 0~3600.

Reply to new REGISTER messages with time difference. This item should work
with SIP Force Expires. For example, if SIP Force Expires is set 1800 seconds
SIP Expires Max
and this item is set 600 seconds, the value of Expires in the 200ok message which
Deviation
is returned by IPPBX upon successful registration will be a random value within the
range of 1200s-2400s. By default, it is 0.
It is null by default, which means not to verify the UserAgent field in the Register
message. When it is not null, a SIP extension can register successfully as long as
UserAgent Filter
the UserAgent field in the Register message sent by the SIP extension is consistant
in the prefix.
Set whether to rewrite the contact port, or rewrite both the contact IP and port. This
SIP Force Contact
function will not take effect until the registration is refreshed. It is null by default.
Call Settings Description
Set the maximum ringing duration in seconds for every call of this extension. The
Call Timeout
default value is 30s.
Set the maximum call duration in seconds for every call of this extension, the call
Max Call Duration will be terminated once it exceeds the time. This item is only valid for calling
external
numbers. The default, value is 6000s.
Wait DialTone Time
The maximum time to wait for the dial tone, calculated by ms.Thedefault value is 0.
Out
Wait Dial Timeout, with the default value of 10000 in milliseconds. The value range
Wait Dial Timeout
is 6000-15000ms.
Digit Timeout, with the default value of 2000 in milliseconds. The value range is
Digit Timeout
1000-10000ms.

Outbound When this feature is set to True, this extension cannot call out except for emergency
Restriction numbers. The default setting is False.

Page 12
 When this feature is enabled, the remote SIP trunk devices can use this extension
and its password to register to this IPPBX and call in without any configuration. You
Extension Trunk
can find this extension in the outbound trunk list and select it as a trunk to call
out. The default setting is False.
CallerID name of this extension trunk displayed in an outbound call once the
Outbound
Extension Trunk feature is enabled, having a higher priority than similar settings in
CallerIdName
Extensions.
CallerID number of this extension trunk displayed in an outbound call once the
Outbound
Extension Trunk feature is enabled, having a higher priority than similar settings in
CallerIdNumber
Extensions.
Set the call permission of an extension, four options available:
No Call: Block any calls from the extension.
Internal Call: Only internal calls are allowed
Call Permission
Local Call: Allow the calls without 0 as the start number
Long-distance Call: Allow the calls with only one 0 at the beginning.
International Call (default): Allow the calls with two 0 at the beginning.
Hotline Settings Description
After FXS hookoff, if you do not have any operations, it will call the hotline number
Hotline Number
which is 2 to 30bits in length. By default, it is null.
Hotline timeout, with the default value of 3000 in milliseconds. The value range is
Hotline Timeout
0~10000(ms) and this value must be less than Wait Dial Timeout.
Other Settings Description
Set the volume in the direction from the SIP phone to the SIP port. The value range
RX Volume
is -4~4 and the default value is 0.
Set the volume in the direction from the SIP port to the SIP phone. The value range
TX Volume
is -4~4 and the default value is 0.
FXS Settings Description
Set the minimum amount of time, in milliseconds, that a hook flash must remain
Min Flash Detection depressed in order for the system to consider it as a valid flash event. The default
value is 300ms.
Set the maximum amount of time, in milliseconds, that a hook flash must remain
Max Flash Detection depressed in order for the system to consider it as a valid flash event. The default
value is 1000ms.
Set the volume in the direction from the analog phone to the FXS port. The value
RX Volume
range is -7~7 and the default value is 0.
Set the volume in the direction from the FXS port to the analog phone. The value
TX Volume
range is -7~7 and the default value is 0.
Echo Cancellation
The default value is 64ms
Level
Set the length of the in-band DTMF voice to cut. Do not set it too large lest normal
Enable Cut DTMF
voice signals be cut. The default value is 25.
Enable
Enable the DTMF passthrough during the conversation. By default it is unticked.
DTMFPassthrough

Page 13
 Press the hook flash on the analog phone during a call to direct this call to 3-way
Flash Event calling or call forwarding. The corresponding options are 3 Way (default) and Call
Swap.
Set the length of the DTMF tone sent by FXS, calculated by ms. The value range is
DTMF Duration
20-800 and the default value is 100ms.
Set the interval for FXS to send DTMF tones, calculated by ms. The value range is
DTMF Gap
20-800 and the default value is 100ms.
Tone Country Two options available: USA (default) and China.
Call Waiting Enable the Call Waiting feature for this extension. By default it is unticked.

3.3.2. Extension Groups

Item Description
The name of the extension group. It is null by default and must be filled in;
Name
otherwise, the configuration will fail to be saved.
Select one or more extensions to become members of the extension group. It is null
Member
by default and must be filled in; otherwise the configuration will fail to be saved.

3.3.3. Trunks

3.3.3.1. Basic
Item Description
Trunk Type Trunk type, SIP or FXO.
Trunk Name User-defined, consisting of letters and digits.
Record Set whether to save the recording data. The default setting is False.
Enabled Enable or disable the trunk. The default setting is True.
Three options available: UDP, TCP, TLS. TLS goes valid only if it is enabled in SIP
Transport
Settings. The default setting is udp.
Set whether to register the SIP trunk, which is determined by the trunk provider. The
Register
default setting is False.
Profile Two options available: LAN (default).

Trunk IP/Domain IP address or domain name of the SIP trunk plus port number.

Username Username of the registered SIP trunk

Auth Username Used for SIP authentication. In most cases, it is the same with the username.
Password The registration password of the SIP trunk.
Expire Seconds The default value is 800 seconds.
RegFail Retry The default value is 30 seconds.
Keep Inbound In case of unregistration, use the transparent extension as the caller by default; in
CallerID case of registration, use the registered account as the caller by default.
Enable Proxy Support of proxy mode for trunks like IMS. By default, it is unticked.

Page 14
 Outbound CallerId CallerID name of this trunk displayed in an outbound call, having a higher priority
Name than similar settings in Extensions. By default, it is null.
Outbound CallerID CallerID number of this trunk displayed in an outbound call, having a higher priority
Number than similar settings in Extensions. By default, it is null.

3.3.3.2. CODEC
Item Description
Set the RTP codec for SIP trunk outbound calls. G711A, G711U, G729, G722,
PCMU, PCMA, GSM, G723, G726-16, speex@8000h@20i, speex@16000h@20i,
Codec Preferences AMR, PCMA@30i, PCMA@40i, PCMU@30i, PCMU@40i, VP8, VP9, H264, H263,
H263-1998, H263-2000are supported at present. If none is selected, all Codecs in
SDP will be used by default; otherwise, only the selected ones will be assigned.

3.3.3.3. Advance
VoIP Settings Description
Set to null to get called ID from the request field; set to auto to user to get called ID
Get CalledID Type
from the to field. By default, it is null.
* NONE (default): Put the CID information only in the From field;
Send CID Type * Remote-Party-ID: Add the Remote Party-ID field to the CID information;
* P-Asserted-Identify: Add the P-Asserted-Identify field to the CID information.

The interval to send the OPTIONS message to check if this SIP trunk is available,
OPTIONS Interval calculated by second. The default setting null means no sending. By default it is
null, which means not to send.
When this item is set to True, the header field Privacy: id will be added to the
Send Privacy ID
INVITE message. By default, it is set to False.
Set the volume for voice from the SIP line to the IP port. The value ranges from -4 to
TX Volume
4 and the default value is 0.
Set the volume for voice from the IP port to the SIP line. The value ranges from -4 to
RX Volume
4 and the default value is 0.
Use the value of this item to override the UserName field in the From header field
From User
while sending the INVITE message. By default, it is null.
Use the value of this item to override the Domain field in the From header field while
From Domain
sending the INVITE message. By default, it is null.
When you select ‘rewrite-contact-IP’, the host field representing contact is
SIP Force Contact
overwritten by the source IP address.
Use the registered account/transparent extension as the caller. Use the transparent
PAIField Source
extension as the caller by default.
When it is ticked, the RTP stream is encrypted and the certificate is the same as
Enable SRTP
TLS. By default it is unticked.
Other Settings Description
Set the maximum number of concurrent calls for this SIP. The default value is 0
Limit Max Calls
which means no limit.

Page 15
Enable Early
Add the message header P-Early-Session: supported to the SIP message.
Session
Enable Early Media Add the message header P-Early-Media: supported to the SIP message.
User Phone You can add content to the To field of the INVITE message.
Set the maximum response time of the trunk for a call out from it. The default value
Call Timout
is 30s.
Dial Number Identification Service is used to identify which trunk a call comes in. It
allows users to define the display name of an incoming call instead of the called
DNIS
number so that the phone will display the DNIS name when a call comes in on the
corresponding trunk. It is unticked by default.
DNIS Description
Dial Number Identification Service is used to identify which trunk a call comes in. It
allows users to define the display name of an incoming call instead of the called
DNIS
number so that the phone will display the DNIS name when a call comes in on the
corresponding trunk. It is unticked by default.
DNIS Name The name of the caller ID displayed for the incoming call through this SIP trunk.
The number of the callee ID of the incoming call through this SIP trunk according to
DNIS Number
which users determine the value of DNIS Name.
FXO Description
Set the volume in the direction from the FXO port to the analog phone. The value
TX Volume
range is -7~7 and the default value is 0.
Set the volume in the direction from the analog phone to the FXO port. The value
RX Volume
range is -7~7 and the default value is 0.
Hangup Detection Description
Hangup Detection
Two methods available: Busy Tone (default) and Polarity.
Method
Busy Count Specify how many busy tones to wait for before hangup. The default value is 4.
Busy Freq Set the frequency of busy tones detected. The default value is 450Hz.
Delay Detect Busy Set the delay time before detecting the next busy tone, calculated by 20ms. The
Tone default value is 25.
Busy Tone
Set the cycle to detect the busy tone, calculated by 20ms. The default value is 200.
Detection Cycle
Answer Detection Description
Set whether to use the Polarity method to detect if the remote end picks up the call
and answers.
None (default): Once an FXO outbound number is successfully sent, the call will be
Answer Detection regarded as answered by the callee and the IPPBX will send 200ok message in the
Method direction to the caller.
Polarity: When an FXO outbound number is successfully sent and the polarity
reversal signal is detected as 1 on the line, the call will be regarded as answered by
the callee and the IPPBX will send 200ok message in the direction to the caller.

DID Number Set the DID number for the incoming call through this FXO port. By default it is null.
Caller ID Settings Description

Page 16
 Caller ID Detection Set whether to detect the Caller ID of an incoming call. By default, it is ticked.
Set the minimum time interval for the answer polarity detection and the hang-up
Polarity Delay
polarity detection. The default value is 600ms.
Set the ring count. If Caller ID Detection is enabled, the value must be greater than
On Ring Count
1. The default value is 0.
DTMF Letter Clear Clear the letters A, B, C, D in the CallerID.
Enable Debug Caller
Enable the caller number debugging feature for recording and analyzing the caller.
ID
Set Record Time for
Set the recording timing, after the first ring or ahead of the first ring.
Caller ID
Other Settings Description
Set the length of the in-band DTMF voice to cut. Do not set it too large lest normal
Enable Cut DTMF
voice signals be cut. The default value is 25ms which means not to cut.
Enable
Enable the DTMF passthrough during the conversation. By default it is unticked.
DTMFPassthrough
Set the length of the DTMF tone sent by FXO, calculated by ms. The value range is
DTMF Duration
20-800 and the default value is 100ms.
Set the interval for FXO to send DTMF tones, calculated by ms. The value range is
DTMF Gap
20-800 and the default value is 100ms.
Wait Dialtone
The maximum time to wait for the dial tone, calculated by ms. The default value is 0.
Timeout
The maximum time to delay for the dial, calculated by ms. The default value is 500.
Delay Dial Timeout If you wish to customize, enter the value in the text box directly. The valid range is
greater than or equal to ‘0’.
Echo Cancellation
The default value is 64ms.
Level
Tone Country Two options available: USA (default) and China.
Fast Release upon Set whether to enable this feature for a trunk. It is disabled by default. If enabled,
Inbound Call Failure inbound calls will be released quickly as they are failed.
FXO Flash Time Set the flash time of the FXO port.

3.3.3.4. DOD
Item Description
This feature allows users to set the caller ID and number of associated extensions
DOD displayed when dialing out which have the higher priority than the caller ID and
number configured in basic settings.
DOD Name The caller ID name of an outbound call.
DOD Number The caller ID number of an outbound call.
Bind Extension Determine which extensions are selected to bind the DOD number

3.3.3.5. Adapt Caller ID

Item Description

Page 17
 Adapt the incoming caller ID number by cutting or adding the prefix in order to
Adapt Caller ID
facilitate the use of the callback feature for the SIP extension.
Use regular expression to match.
(1) ^123$: "^" starts matching, "$" ends matching, ^123$ indicates strict matching
(2) 123^(123|456)$: "|" means or, ^(123|456)$ means to match 123 or 456
(3) ^123|456$:Itmeans to match the string beginning with 123 or the string
endingwith 456
(4) ^123[4-6]$: "[]" means to match any of the characters within, [4-6] is equivalent
to [456], ^123[4-6]$ means matching 1234, 1235, 1236
(5) ^123\d$: "\d" means any number from 0 to 9, ^123\d$ is equivalent to
^123[0-9]$
(6) ^123\d+$: "+" means to match one or more characters in front of it,
^123\d+$ means matching at least 4 digit strings beginning with 123, such as:
Match Mode 1234, 12300, ..., 123456789, etc.
(7) ^123\d*$: "*" means to match 0 or more digits in front of it, ^123\d*$ means
matching at least 3 digit strings beginning with 123, such as:
123,1234,12300 ,...,123400000, etc.
(8) ^123: It means any string beginning with 123, such as: 123, 12345, 123abc,
etc.
(9) 123$: It means any string ending with 123, such as: 123, 666123, abc123, etc.
(10) ^123\d{3}$: "\d{3}" means to match 3 digits, ^123\d{3}$ means matching any
6-digit string beginning with 123
(11) ^(13[4-9]|147|15[0-2,7-9]): It indicates to match the string beginning with 134,
135, 136, 137, 138, 139, 147, 150, 151, 152, 157, 158, 159
(12) .*: "." represents any character, .* means to match any string
Strip Remove the prefix of an incoming call number.
Prepend Add the prefix content after removing the prefix.

3.4. Call Control

3.4.1. Inbound Routes

Item Description
User-defined name of this inbound route. It must be filled in; otherwise the
Name
configuration will fail to be saved.
Enabled Set whether to enable this route. The default setting is True.
Use regular expression to match.
(1) ^123$: "^" starts matching, "$" ends matching, ^123$ indicates strict matching
(2) 123^(123|456)$: "|" means or, ^(123|456)$ means to match 123 or 456
DID Pattern (3) ^123|456$: Itmeans to match the string beginning with 123 or the string
ending with 456
(4) ^123[4-6]$: "[]" means to match any of the characters within, [4-6] is equivalent
to [456], ^123[4-6]$ means matching 1234, 1235, 1236

Page 18
 (5) ^123\d$: "\d" means any number from 0 to 9, ^123\d$ is equivalent to
^123[0-9]$
(6) ^123\d+$: "+" means to match one or more characters in front of it,
^123\d+$ means matching at least 4 digit strings beginning with 123, such as:
1234, 12300, ..., 123456789, etc.
(7) ^123\d*$: "*" means to match 0 or more digits in front of it, ^123\d*$ means
matching at least 3 digit strings beginning with 123, such as:
123,1234,12300 ,...,123400000, etc.
(8) ^123: It means any string beginning with 123, such as: 123, 12345, 123abc,
etc.
(9) 123$: It means any string ending with 123, such as: 123, 666123, abc123, etc.
(10) ^123\d{3}$: "\d{3}" means to match 3 digits, ^123\d{3}$ means matching any
6-digit string beginning with 123
(11) ^(13[4-9]|147|15[0-2,7-9]): It indicates to match the string beginning with 134,
135, 136, 137, 138, 139, 147, 150, 151, 152, 157, 158, 159
(12) .*: "." represents any character, .* means to match any string
Set the DID number to a group of consecutive extension numbers. Please fill in the
DID number range, select "Destination" as "Extension Range" and fill in the
corresponding extension range. For example, DID range: 5503301-5503305,
extension range: 1001-1005.

Caller ID Pattern Same as the item DID Pattern. By default, it is null.

Multiple options available, such as Extensions, IVR Menus, Ring Groups,
Destination Conference Rooms, Call Center, etc. By default it is null and must be filled in;
Otherwise, the configuration will fail to be saved.
Enable Inband
Enable the inband DTMF detection, and the default setting is False.
DTMF Detection
Set whether to enable the fax detection. The default setting is False.
Enable Fax
*False: Neither detect Fax tone nor send Fax.
Detection
*True: Proceed to send Fax if Fax tone detected.

In case the fax detection is enabled and the property of the SDP field in the INVITE
Fax Destination message is detected as fax, it is necessary to set a route to the corresponding fax
destination. By default, it is null.
Local: When an incoming call to the trunk is transferred to the extension and the
extension is ringing, the IPPBX will send the 183 messages well as the ringback
tone to the calling party.
Send Ring Tone Remote: When an incoming call to the trunk is transferred to the extension, the
IPPBX will transmit the 180/183 message and the ringback tone from the extension
transparently to the calling party.
The default setting is Remote.
Enable Time The feature is disabled by default. Once enabled, it is required to set a destination
Condition corresponding to this time condition.
Distinctive Ring Send the INVITE message with the Alert-Info header field to the called extension to

Page 19
 Tone let it select different ring tone files based on the Alert-Info header field. By default, it
is null.
Used to adjust the priority of multiple inbound routes. The default value is 100. The
Priority
smaller the value, the higher the routing priority.
Select the trunks that can use this route. It must be filled in; otherwise, the
Member Trunks
configuration will fail to be saved.
If you enable this setting, when the user’s mobile number dials into the system, the
Enable Mobility phone will have the same user permission as the desktop extension. So, the
Extension mobile number will be able to connect with the other extension, dial out with the
trunk and play voicemail.

Enable T.38 Whether to enable the T.38 feature. The default setting is No.

3.4.2. Outbound Routes

Item Description
User-defined name of this outbound route. It must be filled in; otherwise the
Name
configuration will fail to be saved.
Enabled Set whether to enable this route. The default setting is True.
Use regular expression to match.
(1) ^123$: "^" starts matching, "$" ends matching, ^123$ indicates strict matching
(2) 123^(123|456)$: "|" means or, ^(123|456)$ means to match 123 or 456
(3) ^123|456$: It means to match the string beginning with 123 or the string
ending with 456
(4) ^123[4-6]$: "[]" means to match any of the characters within, [4-6] is equivalent
to [456], ^123[4-6]$ means matching 1234, 1235, 1236
(5) ^123\d$: "\d" means any number from 0 to 9, ^123\d$ is equivalent to
^123[0-9]$
(6) ^123\d+$: "+" means to match one or more characters in front of it,
^123\d+$ means matching at least 4 digit strings beginning with 123, such as:
Dial Patterns 1234, 12300, ..., 123456789, etc.
(7) ^123\d*$: "*" means to match 0 or more digits in front of it, ^123\d*$ means
matching at least 3 digit strings beginning with 123, such as: 123,1234,12300
,….123400000, etc.
(8) ^123: It means any string beginning with 123, such as: 123, 12345, 123abc,
etc.
(9) 123$: It means any string ending with 123, such as: 123, 666123, abc123, etc.
(10) ^123\d{3}$: "\d{3}" means to match 3 digits, ^123\d{3}$ means matching any
6-digit string beginning with 123
(11) ^(13[4-9]|147|15[0-2,7-9]): It indicates to match the string beginning with 134,
135, 136, 137, 138, 139, 147, 150, 151, 152, 157, 158, 159
(12) .*: "." represents any character, .* means to match any string
Strip The number of digits to be removed from the prefix. By default, it is null.
Prepend The digits to be added to the prefix. The default setting is null.

Page 20
Suffix The digits to be added to the suffix. The default setting is null.
Delay The delay time before dial, calculated by ms. The default setting is null.
Add member extensions for controlling the outbound call authority. Only those
Member Extensions extensions selected have the authority to use this route. It must be filled in;
otherwise the configuration will fail to be saved.
Select the trunks that can use this route. It must be filled in; otherwise the
Member Trunks
configuration will fail to be saved.
SIP Code Set the route to the next trunk or route when receiving a specific SIP cause code.
Set the route to the next trunk or route when receiving a specific hangup cause, and
Hangup Cause if the SIP code was set at the same time, the SIP code could not work. By default,
it is null.
If enabled, when the route is successfully matched and the call is not established
Next Route
normally, the next route will continue to be matched. By default, it is ticked.
Set if you need a password for using this outbound route. The default setting is
none.
*None: The call goes out directly
*Pin List: The gateway will require Password for outgoing calls, and will check the
Password
entered PIN with the selected PIN list in Call Features - Pin Numbers. The call will
be preceded while the entered PIN matches any in the PIN list.
*Single Pin: Manually set password .The gateway will require Password for
outgoing calls, and the call will be preceded only if the entered PIN is correct.
Round robin with memory, remember which trunk was used last time, and then use
Rrmemory Hunt
the next available trunk to call out.
Description Description of the outbound route. By default, it is null.
Used to adjust the priority of multiple outbound routes. Smaller Number means
Priority higher Priority. The default value is 1000. The smaller the value, the higher the
routing priority.

Set which time period to use this route. Untick any option by default, which means
Time Condition
no time limits on outbound calls.

Page 21
3.4.3. Outbound Restrictions
Item Description
Name of this user-defined outbound restriction. It must be filled in; otherwise the
Name
configuration will fail to be saved.
Time Limit Set a time limit for calls. The default value is 5 minutes.
Set how many calls are allowed in the limited time. For example, if Time Limit is set
Number of Calls
to 5 minutes and this item is set to 5, it means the designated extension can only
Limit
make 5 calls in 5 minutes. When this extension makes the 6th call, it will be locked.
The setting of True means the designated extension can make more calls after the
Auto Cancel
time limit even if it is locked; the setting of False means this extension, once it is
Restriction
locked, cannot make outbound calls any more until it is unlocked manually.
Enabled Set whether to enable this outbound restriction rule. The default setting is True.
Select the extensions that use this restriction rule. It must be filled in; otherwise the
Member Extensions
configuration will fail to be saved.

3.4.4. AutoCLIP
Item Description
AutoCLIP can redirect call to original extension. The IPPBX automatically stores
information about outgoing calls to the AutoCLIP routing table. When the same
AutoCLIP
person calls back, the call will be routed directly to the original extension that made
the former mentioned outgoing call.

View AutoCLIP List A list of extension outbound calls.

Delete Used Records If enabled, when an AutoCLIP record is matched, it will be automatically deleted
afterwards. By default, it is unticked.

Set how long each record will be kept in the AutoCLIP list. The default value is 8
Record Keep Time
hours.
Only Keep Missed If enabled, the system will only keep records of outbound calls that are not
Call Records answered by the called party in the AutoCLIP list. By default, it is ticked.
Match Outgoing If enabled, only the calls that come in through the same trunk as the last call go out
Trunk from will match against the AutoCLIP list. By default, it is ticked.

Record PSTN Trunk If enabled, calls that go out through PSTN will be recorded to the AutoCLIP list. By
default, it is ticked.

Define how many digits from the last digit of the incoming call number will be used to
Digits Match match the AutoCLIP record. If the number has fewer digits than the value defined
here, it will be matched in full length. The default value is 7.

Enabled Set whether to enable the AutoClip routing. The default setting is False.

Member Trunks Select the trunk on which outgoing calls will be recorded. It is required; otherwise
the configuration will fail to be saved.

Page 22
3.4.5. CC Routes
Item Description
When the extension is busy, the call will be recorded. After the callback interval, the
CC Routes
call will be dialed back.
View the list of calls which are recorded upon the extension is busy and need to be
View CC List
dialed back.
CC Interval Time The callback interval for calls in the CC record. The default value is 1 minute.
Record Keep Time The time to keep a CC record. The default value is 8 hours.
Enabled Set whether to enable the CC routes. By default, it is False.
Member Extensions Add the extensions which have the authority to control the CC routes.

ps: When the extension calls back the caller and the caller picks up the call, it will hear the
prompt ‘This is a call back, please do not hang up’ and then call back the extension. If the call
comes from the FXO port, the caller may not hear the prompt or the prompt may be
incomplete. In such case, it is recommended to turn on the Polarity Reversal function for the
FXO port.

3.4.6. Time Condition

Item Description
It can be set for such features as outbound routes, inbound routes, call forwarding,
Time Condition
and follow me.

User-defined name of a time condition. It must be filled in; otherwise the

Name
configuration will fail to be saved.

Type Three options available: WorkTime (default), Holiday, Custom.

Define worktime and custom conditions necessary to execute the destination
Settings
selection.
Advance If ticked, more settings will appear for you. By default, it is unticked.
Multiple times allowed to set, including day of week, hour and minute by default. If
WorkTime
you need to set year, month, day of month, tick the above item Advance.
Multiple times allowed to set, including year, month and day of month by default. If
Holiday
you need to set day of week, tick the above item Advance.
Multiple times allowed to set, including month, day of month, week of month, day of
Custom
week, hour, minute, as well as exclude holiday.

Page 23
3.5. Call Features

3.5.1. IVR

3.5.1.1. IVR
Basic Description
User-defined IVR name. It must be filled in; otherwise, the configuration will fail to be
Name
saved.
The extension number that can be routed to this IVR, with the default value range of
6500~6599 which can be modified in ‘PBX->Preference->Extension Preferences’. It
IVR Number
is null by default and must be filled in; otherwise, the configuration will fail to
be saved.

It is played as the first prompt for entering the IVR menu. The default setting is
Greet Long
Default.
It is played when the user doesn’t enter any key or enters a wrong key. By default, it
Greet Short
is null.
Response Timeout The time waiting for a digit input after prompt. The default value is 10000ms.
The maximum time between your entering of two adjacent DTMF digits. The default
Inter-Digit Timeout
value is 3000ms.
Max Timeouts Maximum number of timeouts before exit. The default value is 3.
Password Set the IVR password. The default value is 0000.
Max Failures Maximum number of retries before exit. The default value is 3.
Digit Length Maximum number of digits allowed for the caller ID. The default value is 4.
Enabled Set whether to use the IVR. By default, it is True.
Direct Extension Set whether the user can dial directly to extensions after hearing the IVR prompt.
FXO Flash Transfer Set whether to allow the current FXO to flash transfer. By default, it is False.
Set whether the user can dial directly out after hearing the IVR prompt. By default,
Direct Outbound
it is unticked.
Advanced Description
Invalid Sound The prompt played in case of invalid keypress. The default setting is Default.
Exit Sound The prompt played upon exiting the IVR menu. The default setting is Default.
Exit Action The destination selected to enter after exiting the IVR menu. By default, it is null.
Caller ID Name The prefix of the caller ID name sent upon the call passing from IVR to an internal
Prefix extension. By default, it is null.

Page 24
 The ring back tone the caller will hear upon the call passing from IVR to an internal
Ring Back
extension. The default setting is Default.
Key Press Event Description
Keypress Event Set the destinations respectively according to 0~9, *, # and invalid keys.

3.5.1.2. Conference
3.5.1.2.1. Basic
Item Description
User-defined name of a conference room. It must be filled in; otherwise the
Room Name
configuration will fail to be saved.
The number dialed to reach this conference room, with the default value range of
Conference Center 6400~6499 which can be modified in ‘PBX->Preference->Extension Preferences’.
Number It is null by default and must be filled in; otherwise the configuration will fail to be
saved.
The greeting played upon joining this conference room. The default setting is
Greeting
Default.
Schedule Set the start and end time for this conference room.
Set whether a password is needed for entering this conference room. The default
Pin
setting is True.
Record Set whether to enable the recording. By default, it is False.
Max Members The maximum number of members allowed in this conference room.
If set to True, other members will hear prompts upon a member enters or exits this
Announce conference room; if set to False, there will be no prompt for a member’s entering or
exiting. The default setting is False.
Enabled Sets whether to use the conference room. The default setting is True.
Specify the moderator extension for this conference. It must be filled in; otherwise
Moderator Member
the configuration will fail to be saved.
3.5.1.2.2. Advanced Settings
Participant Settings Description
If set to True, the participants could not hear each other until the moderator joins the
Wait for Moderator
conference. The default setting is True.
If set to True, you will hear a prompt ‘Please say your name’ upon you enter a
Say your name conference room, and other members will hear a prompt ‘xxx enters the conference’
upon you successfully join in the conference. The default setting is True.
If set to True, the participants expect for the moderator are not allowed to speak in
Mute Participant
this conference room. The default setting is False.
If set to True, all participants could press *0 to invite other users to enter this
Allow Participant to conference room, press *1 to launch an invitation with confirmation request and
Invite press *2 to kick the member they invited out of this room. The administrator could
press *3 to kick out all participants in the conference. The default setting is True.
3.5.1.2.3. Appointment Meeting Settings
Item Description

Page 25
 If set to Yes, Appointment Meeting will be enabled; if set to No, Appointment
Enable Appointment
Meeting will be disabled. The default setting is No.
Meeting Group
Choose the members who will be invited to the conference meeting.
Name
Custom Date To custom the date of Appointment Meeting. The default setting is Yes.
Date Choose the date of Appointment Meeting.
Time Choose the time of Appointment Meeting.

3.5.1.3. Call Queue

3.5.1.3.1. Basic
Item Description
User-defined name of a call center queue. It must be filled in; otherwise the
Queue Name
configuration will fail to be saved.
The number dialed to reach this call center queue, with the default value range of
6700~6799 which can be modified in ‘PBX->Preference->Extension Preferences’.
Queue Number
It is null by default and must be filled in; otherwise the configuration will fail to be
saved.
Set whether a password is needed for dynamic agents to enter this queue. The
Pin
default setting is False.
Set the password for dynamic agents to enter this call center queue. It is null by
Agent Password
default and must be filled in; otherwise the configuration will fail to be saved.
Ring All: All available agents ring.
Longest Idle Agent (default): The agent keeping idle for the longest time rings first.
Round Robin: All available agents ring in rotation.
Random: All available agents ring randomly.
Ring Strategy
Agent with Least Talk Time: The agent whose total call time is shortest rings first.
Agent with Fewest Calls: The agent with fewest calls rings first.
Top Down: The agents ring from top to down in the order already configured.
Sequentially by Agent Order: The agents ring in the order of their numbers.
Agent Call Timeout The maximum time for each agent to ring. The default value is 15 seconds.
The allowed number of consecutive unanswered calls. 0 means no limit and the
Max No Answer
default value is 0.
The maximum time a caller can wait in a queue before being pulled out, calculated
Max Wait Time
by second. 0 means no time limit.
Discard Abandoned Set the recovery time from disposal.
Select the destination to enter when the call in the queue doesn’t be answered in
Timeout Action
the maximum waiting time. By default, it is null.
Agent Answer
Announcement played upon the agent answers the call. The default setting is null.
Announce
The interval time between the failed and new calls of an agent. The default value is
Agent Retry Time
30 seconds.
Wrap Up Time The interval time between the answer of an incoming call and the allocation of a

Page 26
 new one.
Max Queue Length Set how many callers are allowed in the queue.
Caller ID Name The prefix of a caller ID name sent when the queue allocates a call to the agent. By
Prefix default it is null.
Transfer to the destination when the number of calling parties in the queue exceeds
Overflow Action
the upper limit.
Agents Initial Status Set the initial status of the static agents.
Alert Info Set the content of the Alert-Info field. By default, it is null.
Agents Set one or several extensions to be the fixed station of the current queue.
3.5.1.3.2. Caller Experience Settings
Item Description
Select the music on hold to play when the caller enters this queue. The default
Music on Hold
setting is Default.
Join When No Agent If enabled, callers can join a queue that has no agents. By default, it is unticked.
Max Wait Time with The maximum waiting time for a caller in the queue that has no agents. The default
No Agent value is 90s.
Announcement played to callers upon joining the queue. The default setting is
Join Announce
Default.
Queue Busy Set whether to assign incoming calls to other stations if the current station is already
Resume Offer in call. The default setting is True.
Join Announce
Set the playtime of Join Announce. The default value is 0 which means it is invalid,
Playtime
Set the way to answer. When it is set to Answer, 180 will be returned; when it is set
Answer Type
to Pre-answer, 183 will be returned. The default setting is Answer.
When the calling party enters the queue, the system prompt tone will be heard
Transfer Prompt periodically. If the agent rings, the prompt tone will be played. For example: "Call
forwarding, please wait for a moment". The default setting is empty.
Caller Position
Description
Announcements
Announce Position Announce the current position of the caller in the queue. By default, it is ticked.
Announce Hold
Announce how long the caller shall wait in the queue. By default, it is ticked.
Time
The average call length estimated by users based on actual situations, used to
Call Duration
calculate the waiting time for the caller. The default value is 60s.
Announce Set how often to announce the queue position and the hold time. The default value
Frequency is 30s.
Periodic
Description
Announcements
The system prompt that will be played periodically to callers in the queue, such as
Announce Sound ‘All agents are busy. Please wait a minute. To leave a message, press 1; to end the
call, just hang up’. The default setting is Default.
Announce How often the system prompt is played. The default value is 0s.

Page 27
 Frequency
Busy Callback Description
When this feature is enabled, the caller can choose to hang up the call while
hearing a corresponding voice prompt and this call still waits in line. Then once it is
Enable Busy
the caller’s turn to transfer the call to an agent, IPPBX will start a call to this agent
Callback
and wait for answers before dialing back to the caller to establish a connection. The
default setting is False.
Agent Busy
Select a voice file as the prompt for agent busy. The default setting is Default.
Announce
Agent Busy Press this key to enter the flow of dialing back upon agent busy. The default value is
Callback Key 2.
3.5.1.3.3. Key Press Event
Key Events Description
Option The keys that might be pressed after the caller hears the system prompt.
Destination The destination the call will be transferred to after the caller’s keypress.

3.5.1.4. Pickup Group

Item Description
User-defined name of an intercept group. Users can set intercept groups by service
Name requirements, facilitating the members in a group to answer calls for each other. It
must be filled in; otherwise, the configuration will fail to be saved.
Select members for this group. It is null by default and must be filled in; otherwise,
Member
the configuration will fail to be saved.

3.5.1.5. Ring Group

Item Description
User-defined name of a ring group. It must be filled in; otherwise the configuration
Name
will fail to be saved.
The number dialed to reach this ring group, with the default value range of
6200~6299 which can be modified in ‘PBX->Preference->Extension Preferences’.
Ring Group Number
It is null by default and must be filled in; otherwise the configuration will fail to be
saved.
Three options available: Simultaneous, Sequence, Random.
*Simultaneous (default): All extensions ring at the same time.
Ring Strategy
*Sequence: Ring one by one. Timeout by Second.
*Random: Random select extensions, none-repetitive.

Select the destination to enter when agents in this ring group are all not answered.
Timeout Destination
By default it is null.
The timeout time to ring next extension, and also the timeout time to enter Timeout
Ring Timeout(s)
Destination if all extensions are unavailable. The default value is 30s.
Enabled Set the status of the ring group. The default setting is True.
Alert Info Set the content of the Alert-Info field. By default, it is null.

Page 28
 Ring Back Scheme The ringback tone sent to the caller. The default setting is us-ring.
CID Name Prefix The prefix of a caller ID name sent to the extension. By default, it is null.
Extension Answer If set to Yes, the extension user will hear the following prompts upon picking up the
Confirm call: Press 1 to answer; press 2 to reject. The default setting is No.
Select members for this group. It is null by default and must be filled in; otherwise,
Member
the configuration will fail to be saved.

3.5.1.6. Blocked List

Item Description
User-defined name of a blacklist. It must be filled in; otherwise, the configuration
Name
will fail to be saved.
Set the mode to match the caller number coming in through the trunk with the
Match Mode
blacklist, two options available: Exact Match(default) and Regex Match.
Blocked List
Set the direction of the blocked list. The default setting is Inbound.
Direction
Blocked List
An exact number in the blacklist.
Number
Fill in following the rule of Regular Expression.
(1) ^123$: "^" starts matching, "$" ends matching, ^123$ indicates strict matching
(2) 123^(123|456)$: "|" means or, ^(123|456)$ means to match 123 or 456
(3) ^123|456$: It means to match the string beginning with 123 or the string
ending with 456
(4) ^123[4-6]$: "[]" means to match any of the characters within, [4-6] is equivalent
to [456], ^123[4-6]$ means matching 1234, 1235, 1236
(5) ^123\d$: "\d" means any number from 0 to 9, ^123\d$ is equivalent to
^123[0-9]$
(6) ^123\d+$: "+" means to match one or more characters in front of it,
^123\d+$ means matching at least 4 digit strings beginning with 123, such as:
Regular Expression 1234, 12300, ..., 123456789, etc.
(7) ^123\d*$: "*" means to match 0 or more digits in front of it, ^123\d*$ means
matching at least 3 digit strings beginning with 123, such as:
123,1234,12300 ,...,123400000, etc.
(8) ^123: It means any string beginning with 123, such as: 123, 12345, 123abc,
etc.
(9) 123$: It means any string ending with 123, such as: 123, 666123, abc123, etc.
(10) ^123\d{3}$: "\d{3}" means to match 3 digits, ^123\d{3}$ means matching any
6-digit string beginning with 123
(11) ^(13[4-9]|147|15[0-2,7-9]): It indicates to match the string beginning with 134,
135, 136, 137, 138, 139, 147, 150, 151, 152, 157, 158, 159
(12) .*: "." represents any character, .* means to match any string
Enabled Set whether to enable the black list feature. By default, it is True.

Page 29
3.5.1.7. PIN Numbers
Item Description
User-defined name of a PIN number. It must be filled in; otherwise the configuration
Name
will fail to be saved.
Multiple PIN numbers are allowed and should be separated by ‘,’. This feature is
used for such applications as conference, outbound routes which require entering
PIN List
the PIN number to verify authorities. It is null by default and must be filled in;
Otherwise, the configuration will fail to be saved.
Set whether to enable or disable the PIN list. It is null by default and must be filled
Enabled
in; otherwise the configuration will fail to be saved.

3.5.1.8. Speed Dial

Item Description
User-defined name of a speed dial, which must be unique. It is null by default and
Name
must be filled in; otherwise the configuration will fail to be saved.
Number of a speed dial, unique. It is null by default and must be filled in; otherwise,
Speed Dial Number
the configuration will fail to be saved.
Destination number that the speed dial number corresponds to. It is null by default
Destination
and must be filled in; otherwise the configuration will fail to be saved.

3.5.1.9. Paging
Item Description
User-defined name of a call paging, which shall be unique. It is null by default and
Name
must be filled in; otherwise the configuration will fail to be saved.
Number of a call paging. The default value range is 6300~6399.It is null by default
Number
and must be filled in; otherwise, the configuration will fail to be saved.
Type Two options available: Unilateralism(default) and Bidirectional.
The prefix of a caller ID name of the call started by the call paging. It is null by
CallerID Name Prefix
default.
Select members for this group. It is null by default and must be filled in; otherwise,
Member
the configuration will fail to be saved.

3.5.1.10. DISA
Item Description
User-defined name of a DISA, which must be unique. It is null by default and must
Name
be filled in; otherwise the configuration will fail to be saved.
Response Timeout The maximum time waiting for the caller to press digits after prompt.
Digit Timeout The maximum time permitted between two digits in dialing an extension number.
Second Dial Set whether to enable the two-stage dial. The default setting is Enable.
Three options available: None, Single Pin and Pin List. If set to Single Pin or Pin
PIN Type
List, the caller in DISA will hear the prompt for entering a password before inputting

Page 30
 the callee number to dial.
Select an outbound route for DISA call out. It is null by default and must be filled in;
Outbound Routes
otherwise, the configuration will fail to be saved.

3.5.1.11. Call Back

Item Description
User-defined name of a callback, which must be unique. It is null by default and
Name
must be filled in; otherwise the configuration will fail to be saved.
Delay The delay time to call back after rejecting an incoming call.
Set how many digits will be stripped from the call number before the callback is
Strip
placed. It is null by default.
Set the digits to prefix the callback number before the callback is placed. It is null
Prepend
by default.
The destination which the callback will direct the call to. It is null by default and must
Destination
be filled in; otherwise the configuration will fail to be saved.
*Auto (default)
Through *From Come in
*Select SIP Trunk

3.5.1.12. Emergencies
Item Description
The emergency number users fill in by actual requirements, such as 110, 911. It is
Emergency Number
null by default and must be filled in; otherwise, the configuration will fail to be saved.
Choose trunks for dialing the emergency number. All extensions can make
emergency calls through these trunks regardless of the Time Condition setting.
Trunk When all the trunks are busy, the system will terminate an ongoing call to make sure
the emergency call can be put through. The default setting is None.
When an emergency number is dialed, the system will make a notification call to the
selected extension with a prompt. Multiple extensions are allowed. The default
Announce
setting is None.

Page 31
 3.5.1.13. Multicast
Multicast Type The default type is Triggle Multicast.
Multicast Name The user-defined name of the multicast which must be unique.
Multicast Number The multicast number, with the default range of 6800~6899.
The multicast address, just keep one multicast address at the specific time. The
Multicast Address
default setting is 224.168.168.168.
The multicast port, just keep one multicast port at the specific time. The default
Multicast Port
value is 34567.
Prompt Set the multicast prompt tone.
Enabled Set whether to enable the multicast feature. The default setting is Yes.

3.4.4.14. Wakeup Services

Item Description
User-defined name of the wakeup service, unique. It is null by default and must be
Name
filled in; otherwise, the configuration will fail to be saved.
Prompt The alarm prompt. The default setting is Default.
Custom Date User-defined alarm date, including day of week, etc. By default, it is unticked.
Set the year, month and day information. It is null by default and must be filled in;
Date
otherwise, the configuration will fail to be saved.
Time Set the time which is by default 00:00.
Set the time interval for retry. The value range is ≥60 and the default value is 600,
Snooze Time
calculated by second.
Wakeup Times The wakeup times must be within 0 to 20. 0 means unlimited.
Enabled Set whether to enable the wakeup service. The default setting is True.
The extension members that need the wakeup service. It is null by default and must
Wakeup Member
be filled in; otherwise, the configuration will fail to be saved.

3.6. Billing

3.6.1. Extensions
Item Description
Status Show if the extension is available.
Extension Display the extension number.
Name Display the name of the extension user.
Billing Mode Show the billing method.
Total Recharge Display the total extension call charges.
Balance Display the balance of the extension call charge.
Credit Limit Display the extension call charge limit
Payment Show the extension payment method
Operation Use the edit box to edit the extension, and click Top-up to recharge the extension.

Page 32
3.6.2. Rates
Item Description
Description Enter the description.
Order The order in which the rate matches. The smaller the value, the higher the priority.
Match Pattern Match using regular expressions.
Rate Cost deducted per billing cycle.
To match the rate, the length of dialed number must be shorter or equal to this
Number Length
value. ‘0’means matching numbers of any length.
Billing Increment The billing cycle, with the default value of‘60s’.
Time Condition Set the time period for rate matching. The default setting is ‘All’.
Extension Select the extensions that will use this rate.

3.6.3. Top-up History

Item Description
Top-up Date Display the top-up date.
Extension Display the extension number.
Balance before
Show the balance before recharge.
Top-up
Top-up Amount Show the top-up amount.
Top-up Balance Display the balance after recharge.

3.6.4. Statistics
Item Description
Time Range You can query the extension's billing statistics based on the call start and end time.
Screening Filter based on call data.
Statistics Type The statistic type.
Trunk Filter by trunk.
Extension Filter by extension.

Page 33
 3.6.5. Billing Settings
Item Description
Enable Billing Check it to enable the billing function.
Decimal Places The decimal places to retain in calculating the amount.
Insufficient Balance The prompt will be played when the extension do not have sufficient balance for
Prompt calling outbound.
Extension Locked
The prompt will be played when the extension is locked.
Prompt
Low Balance When the balance reaches the threshold, the system will play the ‘Top-up Prompt’.
Threshold The value must be equal to or greater than ‘0’. The default value is ‘0’.
Top-up Prompt The prompt will be played when the balance reaches the threshold.
Hangup when the
When the account balance is less than the credit limit, the call will be hung up
balance is
automatically.
Insufficient

3.7. FCM Setting

3.7.1. SIP Settings
Item Description
Enable Session Enable the timer for a SIP session which should be refreshed in a designated time.
Timer It is ticked by default.
Set the maximum refresh interval for the session timer. The default value is 1800
Session Timeout
seconds.
The content of the User-Agent field which is defined by users. The default setting is
UserAgent
IPPBX.
RTP Range Set the range of the RTP port used by the PBX. The default setting is 16384-32768.
RTP AutoFix Timing RTP AutoFix Timing. The default setting is True.
When it is set to True by default, the PBX will send the options message to all the
Nat Options Ping
terminals which register after NAT to keep the active connection to the terminal.
In order to guarantee the communications Qos, the eight identification bytes in the
DSCP Enabled IP header of the data packet are encoded to classify the service classes and
differentiate the service priorities.
Enable RTPAlways When this feature is enabled, UC will update the RTP port of the local end as long
Adjust as the PTP port changes. The default setting is No.
Trunk Profile Setting Description
Enable By ticking this option, you can create SIP trunks on the LAN port. It is ticked by
External_LAN default.
Enable By ticking this option, you can create SIP trunks on the WAN port. It appears only
External_WAN when the network mode is set to Double or Route. It is ticked by default.
The IP address to be monitored by using the SIP protocol. By default, it is the IP
SIP IP
address of this network port.
SIP Port The port to be monitored by using the SIP protocol. By default, it is 5080.
Public SIP IP The SIP IP used for NAT traversal when the PBX stays in the LAN.
Public RTP IP The RTP IP used for NAT traversal when the PBX stays in the LAN.

Page 34
100rel Enable Add Supported:100rel to the INVITE message.
If this option is ticked, the SIP trunk will support UDP, TCP, TLS at the same time. It
Enable TLS
is unticked by default.
TLS Only If this option is ticked, the calls on this SIP trunk will only support TLS.
TLS SIP Port The default value is 5081.
TLS Version The TLS version used by the SIP trunk. The default value is tlsv1.
The certificate needed in case the PBX works as the client. It will be renamed to
TLS Certificate
agent.pem after it is uploaded.
Extension Profile
Description
Setting
By ticking this option, you can create SIP extensions on the LAN port. It is ticked by
Enable Internal_LAN
default.
Enable By ticking this option, you can create SIP extensions on the WAN port. It appears
Internal_WAN only when the network mode is set to Double or Route. It is ticked by default.
The IP address to be monitored by using the SIP protocol. By default it is the IP
SIP IP
address of this network port.
SIP Port The port to be monitored by using the SIP protocol. By default, it is 5060.
Public SIP IP The SIP IP used for NAT traversal when the PBX stays in the LAN.
Public RTP IP The RTP IP used for NAT traversal when the PBX stays in the LAN.
Enable WebRTC When it is enabled, users are allowed to register to IPPBX using page.
WS Port Set the WS port. By default, it is 5066.
WSS Port Set the WSS port. By default, it is 7443.
If this option is ticked, the SIP extension will support UDP, TCP, TLS at the same
Enable TLS
time. It is unticked by default.
TLS Only TLS If this option is ticked, the calls on this SIP extension will only support TLS.
TLS SIP Port The default value is 5061
TLS Version The TLS version used by the SIP extension. The default value is tlsv1.
Create CA When the PBX works as the Server, the CA certificate is used to generate Client
Certificate and Server certificates, with the filename cafile.pem.
Create Server It is a certificate needed when the PBX works as the Server, with the filename
Certificate agent.pem.
Create Client It is a certificate provided by the PBX for other clients to use, generated by using the
Certificate same CA certificate of the Server certificate, with the filename client.pem.
RTP CODEC Description
Global RTP Codecs as list: G711A, G711U, G729, G722, GSM, G723, G726-16,
RTPCodec speex@8000h@20i, speex@16000h@20i, AMR, PCMA@30i, PCMA@40i,
PCMU@30i, PCMU@40i, VP8, VP9, H264, H263, H263-1998, H263-2000
Certificate Description
Create
Certificate/Downloa Click it to generate and download the certificate.
d Certificate

Page 35
3.7.2. Record Settings

Item Description
Internal Call Being The prompt that will be played to both the caller and the callee before the recording
Recorded Prompt of internal calls. The default setting is None.
Outbound/Inbound
The prompt that will be played to both the caller and the callee before the recording
Calls Being
of outbound/inbound calls. The default setting is None.
Recorded Prompt
Set the timing to start recording, two options available: After Answer and After
Record Start
Media. The default setting is After Answer.
Set the recording mode, two options available: Recording on One
Record Mode
Side and Recording on Both Sides. The default setting is Recording on One Side.
Set the recording direction, three options available: Incoming Recording, Outgoing
Record Direction Recording, Incoming and Outgoing Recording. The default setting is Incoming
Recording.
Set the sampling frequency of the recording, 8000 or 16000.The default value is
Record Sample Rate
8000.
Recording File
Set the recording file format, WAV or MP3. The default format is WAV.
Format
Record Trunks Select trunks on which the calls will be recorded. By default, it is null.
Record Extensions Select extensions on which the calls will be recorded. By default, it is null.
Record Conferences Select conference rooms in which the calls will be recorded. By default, it is null.

3.7.3. Voice Prompts

3.7.3.1. Voice Prompts

Item Description
The music catalog to play when a call is being held. The default setting is default
Music On Hold
catalog.
Play Call If enabled, the system will play a prompt before transferring a call. By default, it is
Forwarding Prompt unticked.
Set what to play when a call is being held during call forwarding. The default setting
Music On Hold
is Music On Hold.

3.7.3.2. System Prompt

Item Description
Upload System The supported compression format is zip. Please make sure of the integrity of voice
Prompts packages to guarantee the normal use.
Display all the voice packages in IPPBX and allow you to select one as the system
Prompts List
prompt.
Language Three options are available: English (default), Chinese and Turkish.

Page 36
 3.7.3.3. Music on Hold
Item Description
Select a catalogue of music on hold or press the following button + to create a new
Catalogue
catalog.
File Path Select a new music file and upload it to the list.
File List Music files in the list can be played or removed.

3.7.3.4. Custom Prompt

Item Description
The file to be uploaded should be: 8000Hz sampling rate, 16bit, single channel, wav
Upload
or mp3 format.
Define the name of a wav file, select an extension to record, then click the
Record RECORD button. When the extension rings, pick up the call and say what you want
to record.

3.7.4. Storage

3.7.4.1. Preference
3.7.4.1.1. Storage Locations
Item Description
A location to store your voicemail. It is Local Flash by default. If you plug TF or USB
Voicemail storage cards to the PBX, or add network disks, there will be more options: TF/SD,
USB or the network disk (user-defined name).
A location to store your recordings. It is Local Flash by default. If you plug TF or
Recordings USB storage cards to the PBX, or add network disks, there will be more options:
TF/SD, USB or the network disk (user-defined name).
A location to store your One Touch Recordings. It is Local Flash by default. If you
OTR plug TF or USB storage cards to the PBX, or add network disks, there will be more
options: TF/SD, USB or the network disk (user-defined name).
A location to store your logs. It is Local Flash by default. If you plug TF or USB
Logs storage cards to the PBX, or add network disks, there will be more options: TF/SD,
USB or the network disk (user-defined name).
3.7.4.1.2. Storage Devices
Item Description
Display the total storage, available size, usage of the local flash card, providing a
LOCAL
reference for storage setting.
Display the total storage, available size, usage of the external TF card, providing a
TF/SD
reference for storage setting.
Display the total storage, available size, usage of the external USB card, providing
USB
a reference for storage setting.
Display the total storage, available size, usage of the added network disk, providing
NETDISK
a reference for storage setting.

Page 37
3.7.4.2. Auto Cleanup
3.7.4.2.1. CDR Auto Cleanup
Item Description
Set the maximum number of CDR that should be retained. The default value is 10

Max Number of CDR and the value 0 means no limit. If the threshold is reached, the oldest CDR will be
deleted.

CDR Preservation Set the maximum number of days when CDR should be retained. The default value

Duration is 0 which means no limitation. If the threshold is reached, the oldest CDR will be
deleted.
Max Number of
Set the maximum number of conference records allowed to save. If it is exceeded,
Conference
the oldest record will be deleted. The default value is 5000, and 0 means no
Sessions
limitation.
3.7.4.2.2. Voicemail and One Touch Recording Auto Cleanup
Item Description
Set the maximum number of voicemail and one touch recording files that should be
Max Number of Files retained respectively for each extension. The default value is 30. If the threshold is
reached, the oldest data will be deleted.
Set the maximum number of days for voicemail and one touch recording files to be
Preservation
retained respectively for each extension. The default value is 0 which means no
Duration
limitation. If the threshold is reached, the oldest data will be deleted.
Set the maximum number of minutes for voicemail and one touch recording files
Files Preservation
To be retained respectively for each extension. The default value is 0 which means
Duration
no limitation. If the threshold is reached, the oldest data will be deleted.
3.7.4.2.3. Recordings Auto Cleanup
Item Description
Set the maximum storage percentage of recording files for the device. The default
Max Usage of
value is 80% and the value range is 30%~90%. If the threshold is reached, the
Device
oldest data will be deleted.

Rec Preservation Set the maximum number of days for recording files to be retained. The default
value is 0 which means no limitation. If the threshold is reached, the oldest data will
Duration be deleted.

3.7.4.2.4. Logs Auto Cleanup

Item Description
Max Size of Total Set the maximum size of logs that can be saved per file. The default value is 50MB.
Logs If the threshold is reached, the oldest data will be deleted.
Logs Preservation Set the maximum number of log files to be saved. The default value is "7", and "0"
Duration means no limit.

Page 38
 3.7.4.2.5. Backups
Item Description
After the information of the FTP server is configured, the recording file will be
Auto Upload FTP
uploaded automatically. The default setting is False.
FTP server address, format: xxx:xxx:xxx:xxx or xxx:xxx:xxx:xxx:xx. It is null by
FTP Address
default and must be filled in; otherwise, the configuration will fail to be saved.
Username User name used on the FTP server
Password Password used on the FTP server
Real Time (default): upload every 5 minutes
Upload Time Timing: Upload at a fixed time every day. If you select this item, you need to set the
startup time and the default setting is 00:00.
Set whether to delete the original recording file after it is uploaded. The default
Delete Source File
setting is False.
FTP Test After the above configurations are set, you can test whether the FTP connection
goes normal.

3.7.4. Preference

Item Description
The maximum time length permitted for a call. The default value is 6000 seconds. 0
Max Duration
means no limit.
The Caller ID that will be displayed on the recipient’s phone. There options
available: Auto, Transferor(default), Transferee.
Example: 1002calling 1003, 1003transfers this call to 1004.
Attended Transfer * Transferor: Show 1003 on the extension 1004.
Caller ID * Transferee: Show 1002 on the extension 1004.
* Auto: When 1003istransferring the call to 1004, the displayed number is 1003;
when 1003 hangs up the call after its successful transfer and1002 is talking to 1004,
the displayed number is 1002.

Extension
Description
Preferences
User Extensions The number range of user extensions. By default, it is 1000~5899.
Ring Group
The number range of user extensions in a ring group. By default, it is 6200~6299.
Extensions
Paging Group
The number range of user extensions in a paging group. By default, it is 6300~6399.
Extensions
Conference The number range of user extensions in a conference room. By default, it is
Extensions 6400~6499.

IVR Extensions The number range of IVR extensions. By default, it is 6500~6599.

The number range of user extensions in a call center queue. By default, it is

Queue Extensions
6700~6799.

Multicast Number The multicast number must be within this range. The default setting is 6800-6899.

Page 39
 API Description
Enable API Enable and disable the API interface. By default, it is disabled.
Username API username. The default value is api.
Password API user password. The default value is api.
Event Information The address for the API to download the recording file. The default setting is the IP
Address address set by the system.
Incoming Call
The timeout period in second for incoming calls controlled by the API.
Control Timeout

3.7.5. Voicemail

Message Options Description

Max Messages per The maximum number of messages to store in a single folder of voicemail. The
Folder default value is 100.
The maximum length of a single piece of message. The default value is 300
Max Message Time
seconds.
Min Message Time The minimum length of a single piece of message. The default value is 3 seconds.
If this option is ticked, you will hear the prompt: The phone you dial is unavailable
Press 5 to leave a
now. Please press 5 to leave your message; if it is unticked, you will hear the
message
prompt: The phone you dial is unavailable now. By default, it is ticked.
Operator Breakout If this option is ticked, you will hear an extra prompt: Press 0 for operator. By
from Voicemail default, it is unticked.
Greeting Options Description
Select the greeting that will be played when the extension is busy. The default
Busy Prompt
setting is Default.
Select the greeting that will be played when the extension is unavailable. The
Unavailable Prompt
default setting is Default.
Playback Options Description
Announce Message If this option is ticked, the extension number of the caller who left the message will
Caller ID be announced before the content of this message. By default, it is unticked.
Announce Message If this option is ticked, the duration of the message will be announced before the
Duration content of this message. By default, it is unticked.
Announce Message If this option is ticked, the arrival time of the message will be announced before the
Arrival Time content of this message. By default, it is unticked.
Click to view the voicemail configuration of all extensions. Then click ‘Messages’
View Voicemail
following each extension to view the exact voicemail messages.

Page 40
3.8. System Setting

3.8.1. Security Strategy

3.8.1.1. Security Strategy
Static Defense Description
Enable Firewall It is ticked by default.
Enable Ping If it is unticked, the ping will be forbidden. By default, it is ticked.
By default, it Is unticked. Add at least one rule that allows TCP to connect to
Drop All
HTTPS:443, telnet:23 before ticking.
The way to add a static security strategy is the same as adding a firewall rule for
Add
Linux.
Auto Defense Description
Enter the port for auto defense. It is null by default and must be filled in; otherwise,
Port
the configuration will fail to be saved.
Protocol Select a protocol for auto defense, including TCP (default) and UDP.
Number of IP The allowed number of packets received within the ‘time interval’. If the amount of
Packets data from a certain IP packet within the ‘time interval’exceeds this threshold, the IP
will be blacklisted. It is null by default and must be filled in; otherwise, the
configuration will fail to be saved.
Time Interval Time interval for receiving packets, calculated by second. It is null by default and
must be filled in; otherwise, the configuration will fail to be saved.
Blacklist Those calls which meet the above set conditions will be blacklisted herein. It can be
manually deleted.
Fail2ban Description
Setup Fail2ban Enable Fail2bna. It is unchecked by default
Ban Time The time in second when it is banned by firewall.
Time Span Within the specified time range (s), after the host reaches the maximum number of
matches, it will be banned.
Max Retry The maximum number of times the host is matched
Blocked List Those meeting the above setting conditions will be listed in the blacklist which can
be deleted manually.

Page 41
3.8.2. Network Settings

3.8.2.1. Basic Settings

Item Description
Hostname The default value is IPPBX.
Three options available: Dual, Bridge, Route. The default mode is Dual.
Dual: Use Both Eth to communicate.
Mode Bridge: Working as switch with LAN address activate.
Route: Working as a router, Only WAN used to communicate, LAN supports DHCP
server in Router Mode.
When the IPPBX is in the Dual network mode, users should make an interface
Default Interface
selection from LAN and WAN. The default setting is WAN.
IPv4 Description

Network Mode Three options available: IP, DHCP, PPPoE, which are the same as the PC in
settings.

3.8.2.2. OPENVPN
Item Description
Enable OpenVPN Choose whether to enable VPN.
Server Address Enter the server address of OpenVPN.
Server Port Enter the server port of OpenVPN. The default value is 1194.
Protocol Select the protocol type. The client and server must use the same setting.
Select the network device. The client and server must use the same setting.
Device Mode TUN: A TUN device is a virtual point-to-point IP link;
TAP: A TAP device is a virtual Ethernet adapter.
Username Specify the username.
Password Specify the password.
Encryption Select the encryption method. The client and server must use the same setting.
Enable or disable compression for data stream. The client and server must use the
Compression
same setting.
CA Certificate Upload a CA certificate
Certificate Upload a client certificate
Key Upload a client key.
TLS Authentication Enable or disable TLS authentication. If enabled, please upload a TA key.

3.8.2.3. Static Routes

Item Description
Add Routes The way to add routes is the same as that for the PC.

Page 42
3.8.3. LDAP Server

3.8.4. Phone Book

Item Description
This address book is only applicable for UCTALK APP, the access port is 10600,
Phone Book
and the uploaded file must be named contacts.txt.
Choose a File Select the address book to upload.
Phone Book
Download a template of the address book.
Template Download

3.8.5. Email Settings

Item Description
The email account which is used to send emails, in the format of [email protected] is
Username
null by default and must be filled in; otherwise, the configuration will fail to be saved.

The login password of the Email account used to send emails. It is null by default
Password
and must be filled in; otherwise, the configuration will fail to be saved.

The display name for the email being sent. It is null by default and must be filled in;
Display Name
otherwise the configuration will fail to be saved.
Only the SMTP server is supported now whose format is smtp.qq.com.It is null by
Send Mail Server
default and must be filled in; otherwise, the configuration will fail to be saved.

The port of the SMTP server, with the default setting of 25. It must be filled in;
Port
otherwise the configuration will fail to be saved.
Enable SSL/TLS Depend on if the mail server requires or not. It is ticked by default.
Whether to enable SSL/TLS for encryption according to the sending server. The
SSL/STARTTLS
default setting is STARTTLS.
After settings are done, click Test Mail to check if the settings are correct. A test
Test Mail
email will be send to the mailbox.

Page 43
3.8.6. Service
Service Description
Set the automatic logout time of the webpage, up to 120 minutes, the default value
Auto Logout Time
is 60, calculated by minute.
Protocol Select the type for webpage access, the default setting is HTTPS.
Port Set the port for webpage access, the default value is 443.
Redirect from Port If it is enabled, the access to Port 80usingtheHTTP protocol will be automatically
80 redirected to the corresponding port of HTTPS. By default, it is ticked.
Set whether to enable Telnet and the corresponding port. By default, it is ticked and
Enable Telnet
the port is 23.
Set whether to enable FTP and the corresponding port. By default, it is ticked and
the port is 21. The FTP login username and password are the same as the admin
Enable FTP
user. After logging in, you can check the recording data under the storage space
such as FLASH, USB mobile hard disk and TF card.
Enable TFTP Set whether to enable TFTP. By default, it is ticked.

3.8.7. Event Setting

3.8.8. Feature Code

Digits Timeout Description
Feature Code Digits The maximum time waiting for the next feature code digit. The default value is
Timeout 5000ms.
Recording Description
One Touch Record The feature code that is used to start call recording. The default code is *2.
Dial the function code+IVR number, you can record the prompt tone, replace the
Agent Free/Busy
welcome word of IVR or restore the configuration. The default code is*10.
Voicemail Description
The feature code that is used to check the voicemail. Press it and enter your
Check Voicemail
password following the prompt. The default code is *97.
Voicemail Main The feature code that is used to access the global menu for voicemail. The default
Menu code is *98.
Voicemail for The feature code that is used to leave a voicemail to specified extensions or forward
Extension an incoming call to an extension’s voicemail directly. The default code is *99.
Transfer Description
Extension A presses this feature code in a call and dials Extension B after hearing
Blind Transfer
the dial tone to transfer the call successfully. The default code is *1.
Extension A presses this feature code in a call, dials Extension B after hearing the
Attended Transfer dial tone, and hangs up the call after communication to transfer the call
successfully. Press # to cancel. The default code is *4.
Attended Transfer In negotiating a transfer, if the destination is not specified, the call will be transferred
Timeout back after the set time. The default value is 15 seconds.
Intercept Description
Group Intercept By pressing this feature code, an extension can answer the incoming call to another

Page 44
 extension in the same intercept group. The default code is *8.
By dialing this feature code plus an extension number, users can answer incoming
Extension Intercept
calls to this extension. The default code is **.
Intercom Description
By dialing this feature code plus an extension number, users can start an intercom
Intercom
call to this extension. The default code is *80.
Call Parking Description
Dial this feature code during a call to put the call on hold and park it at an extension
Call Parking number directed by the system. Any other phone can dial this extension number to
resume the conversation.
Dial this feature code and an extension number to park the call at the extension.
Any other phone can dial this extension number to resume the conversation. The
Directed Call
default feature code is #72.
Parking
Note: If the specified number has been taken, an available number will be assigned
by the system for parking.
By dialing this feature code, Extension A will be parked at another extension
number. Other extensions can dial this extension number to resume the
Park Extension conversation with Extension A. The default feature code is 5900.
Note: If the specified number has been taken, an available number will be assigned
by the system for parking.
Park Extension
The range of extensions where the call can be parked at. The default setting is
Start/Park Extension
5901~5999.
End
The maximum time for an extension allowed to park. The default value is 90
Park Timeout
seconds.
Call Forwarding Description
By dialing this feature code, an extension forwards all calls to its voicemail; by
Enable Forward All
dialing this feature code plus a designated number, an extension forwards all calls
Calls
to this designated number. The default feature code is *72.
Disable Forward All Dial this feature code to disable forwarding of all calls. The default feature code is
Calls *73.
Toggle Forward All Dial this feature code to toggle forwarding of all calls. The default feature code is
Calls *72.
By dialing this feature code, an extension forwards all calls to its voicemail when
Enable Forward
busy; by dialing this feature code plus a designated number, an extension forwards
When Busy
all calls to this designated number when busy. The default feature code is *90.
Disable Forward Dial this feature code to disable call forwarding when busy. The default feature code
When Busy is *91.
By dialing this feature code, an extension forwards all calls to its voicemail when no
Enable Forward No answer; by dialing this feature code plus a designated number, an extension
Answer forwards all calls to this designated number when no answer. The default feature
code is *92.
Disable Forward No Dial this feature code to disable call forwarding when no answer. The default feature

Page 45
Answer code is *93.
DND Description
Enable Do Not Dial this feature code to put the extension into the DND state. The default feature
Disturb code is *77.
Disable Do Not Dial this feature code to take the extension out of the DND state. The default feature
Disturb code is *78.
Toggle Do Not
Dial this feature code to toggle the DND state. The default feature code is *77.
Disturb
Call Monitor Description
Dial this feature code plus an extension number to monitor the extension. If this
Listen feature will work or not is related to the setting of monitor authority. The default
value is *54.
Dial this feature code plus an extension number to monitor the extension and
Whisper whisper to it. If this feature will work or not is related to the setting of monitor
authority. The default value is *55.
Dial this feature code plus an extension number to enter the call of this extension for
Barge-in monitoring. If this feature will work or not is related to the setting of monitor
authority. The default value is *56.
By dialing this feature code in a call, users can disconnect this call forcibly. The
Forcible Hangup
default feature code is *6.
Dial this feature code plus an extension number to monitor the conversation from
the designated extension and the monitor cannot hear the voice from the other
Listen Local end. The default feature code is *93.
Note: To monitor an extension, you need to configure the Monitor Settings for this
extension first.
Dial this feature code plus an extension number to monitor the conversation from
the remote end of the designated extension and the monitor cannot hear the voice
Listen Remote from the local designated extension. The default feature code is *94.
Note: To monitor an extension, you need to configure the Monitor Settings for this
extension first.
By dialing this feature code in a call, users can disconnect this call forcibly on
Force Hangup on
monitor. Pressing this feature code+queue number on extension can disconnect
Monitor
this call forcibly. The default feature code is *64.
Agent Description
By dialing this feature code plus a queue number, the extension can follow the
Agent Login/Logout
prompt to log in and out the queue dynamically. The default feature code is *22.
When the status of the agent in the queue is ‘On Break’, it is set to ‘available’; and
Agent Free/Busy
when the status is ‘available’, it is set to ‘On Break’.
By dialing this feature code plus a queue number, the extension can follow the
Agent Status ID prompt to add an extension to the designated queue or delete it from the queue.
The default feature code is *23.
BlackList Description

Page 46
 By dialing this feature code, the extension can follow the prompt to add a caller Id to
Blacklist Add
the blacklist dynamically. The default feature code is *40.
By dialing this feature code, the extension can follow the prompt to remove a caller
Blacklist Remove
Id from the blacklist dynamically. The default feature code is *41.
Blacklist Add Last By dialing this feature code, the extension will add the latest call number to the
Call blacklist. The default feature code is *42.
Query IP Description
By dialing this feature code, the FXS extension can query such information as the
Query LAN IP
IP address of LAN. The default feature code is *60.
By dialing this feature code, the FXS extension can query such information as the
Query WAN IP
IP address of WAN. The default feature code is *61.
CC Routes Description
When the extension is busy, dial this function key to implement the callback feature.
CC Routes
The default feature code is *7.
Conference Description
Multiparty
The feature of multi-party conference, and the default feature code is *0.
Conference
Custom Status Description
Press the feature code+1 or 2 digit-status code to push the API message. The
Custom Status
default value is *76.

3.8.9. Date Time Settings

Item Description
Current System
Display the current system date and time of the PBX.
Time
Time Zone The default setting is GMT+8:00 (Beijing).
Daylight Saving
Enable it to set the daylight saving time.
Time
Set the date and time manually. Tick the option System Time below and you can
Set up Manually
manually set the time.
Synchronized with Fill in the address or domain name of a NTP server and the PBX will synchronize
NTP Server with it in time automatically.

System Time Set the system time manually.

Enable NTP Server Tick this option to provide the NTP service for other devices.

Page 47
3.9. Maintenance
3.9.1. Upgrade

Item Description
Manual Upgrade Use the upgrading file to upgrade the PBX version manually.

3.9.2. Users
This interface is used for adding WEB users. The default user and its password are both
admin. The admin user can log in to the device through FTP to access the USB, network disk
and local recording folder. The initial password is admin and you can modify it via the web page.

Item Description
Username User-defined, not allowed to be Admin.
Password User-defined.
Confirm Password Confirm your password.
Language Select a language, Chinese or English.
Groups Determine the user’s authority
Enabled Set the status of this account.

3.9.3. User Group

Item Description
By default, an administrator group has the authority to checkstatus, call records and
Admin set recordings, as well as PBX, system and all functional modules. The exact
authority of corresponding functional modules can be set by requirements.
By default, a public group only has the authority to checkstatus and call records, as
Public well as play and query recordings. The exact authority of corresponding functional
modules can be set by requirements.
By default, a user group only has the authority to check status and call records, as
User well as play and query recordings. The exact authority of corresponding functional
modules can be set by requirements.

3.9.4. Reboot

Item Description
Reboot Reboot the IPPBX system.
Auto Reboot Set auto reboot in a day, a week or a month.

Page 48
3.9.5. Backup and Restore

3.9.5.1. Backup
The backup content includes: System Configuration (default), Network Configuration, CDR,
Operation Log Record, Customized Voice Prompt Files (default), SystemVoice Prompt Files.
Users can customize the backup content.

3.9.5.2. Restore
Click the Browse button to select a backup file on your PC to restore your device.
3.9.5.3. Backup Lists
Display all lists of files that have been backed up with the backup time. Here you can select a
backup file to restore.

3.9.6. Factory Reset

Item Description
Restore to the factory settings. You can choose not to restore the network settings.
Factory Reset You should enter the correct verification code for reset, which is randomly
generated

3.9.7. Operation Log

Item Description
Main WEB operations will all be recorded to operation logs which can be queried by
Filter
Username, IP Address, Start and End Date.
The log list will display the operation time, the user who operated, the IP address,
Display
the type of operation as well as the operation details.

3.9.8. PBX LOG

Item Description
Six options available: CONSOLE, INFO, NOTICE, WARNING, ERROR, DEBUG.
Log Level When DEBUG is ticked, you can set subsequently whether to output ‘siptrace’
which is the log of SIP messages.
The system will generate a log file every day which can be downloaded and
Log List
deleted.

3.9.9. Log Viewer

Item Description
The key calls will be recorded in logs. On this interface you can filter those logs and
Log Viewer sort them in descending order, show their line numbers and set their display size so
as to better view the current log information.

Page 49
3.9.10. Trouble Shooting
Item Description
Ethernet Capture
Set filter conditions for network capture, such as SIP only, both SIP and RTP, etc.
Tool
Designate an FXO or FXS port for recording, as well as designate an FXO port to
Port Monitoring Tool
pick up or hang up the call.
IP Ping Test connection of the destination via IP ping.
Trace Route Test the network route and path as well as the response time.
Call Testing Do simple call tests for UC.

3.9.11. Event Log

Item Description
All the logs that are reported by the trigger event will be recorded in the event log.
Event Query
You can query them by ‘Event Type’, ‘Event Name’, and ‘Time’.
The log list shows such detail information as time, event type, event name, and log
Event Display
content.

3.10. Call Detail Records

3.10.1. Call Detail Records
See below for all kinds of query conditions of call records.

Basic Description
Time Range Query CDR according to the start and end times.
Call From Usually, it is the calling party number.
Call To Search the CDR information according to the final destination number.
Direction Three options available: Inbound, Outbound and Local
Call Status Include such options as Answered, Missed, Voicemail, Cancelled, Failed, etc.
Talk Duration Query CDR according to the time length of the call.
Advanced Description
Hangup Cause Query CDR according to the reason why the call ends.
Query CDR according to Mean Opinion Score (MOS) which is a measure of voice
MOS Score
quality.
CID Name Query CDR according to the name of caller identification (CID).
OriginalDestination Query CDR according to the original destination of the caller.
Trunk Name Query CDR according to the used trunk name.
Outbound CallerId
Query CDR according to the calling party number of the outgoing call.
Number
Last Destination Search the CDR information according to the lastest destination.
DID Number Search the CDR information according to the DID number.
Wait Time in Queue Search the CDR information according to the wait time in queue.
Agent Ring Time Search the CDR information according to the agent ring time.
Agent Talk Time Search the CDR information according to the agent talk time.

Page 50
3.10.2. Extension Summary

See below for all kinds of query conditions of call records.

Basic Description
Quick Select Count extension calls based on an approximate time. The default setting is today.
Custom Choose Description
Time Range Count extension calls according to the start and end times.

3.10.3. Conference Recording Session

See below for all kinds of query conditions of conference records.

Basic Description
Start Time/End Time Query conference calls according to the start and end times.
Room Name Name of the conference room.
Conference Center Number of the conference center.
Number

3.11. Value Added Features

3.11.1. Authorization
3.11.1.1. Authorization Info
Item Description
Serial Number Device serial number
Max Sessions Concurrent number. The default value for FCM5404 is 30.
The number of extensions. The default value for FCM5404 is 150.
Max extensions

Features The features supported by this device,

3.11.1.2. Upload Authorization File

Item Description
Manually upload the authorization file to the IPPBX and you can view the latest
Upload
authorized information in ‘Authorization Information’.

3.11.1.3. Clear Authorization

Item Description
Clear Enter your password to do the clearance.

Page 51
3.11.12. Auto Provision

Item Description
This feature is to generate a configuration file of the IP phone. The phone obtains
Auto Provision the configuration file address by sending a paging packet, and downloads the
configuration file automatically.
Manufacturer Select the IP phone manufacturer that generates the phone book.
When the type is Local Phone Book, it cannot be modified; when the type is Remote
Name
Phone Book, the phone book can be obtained by this name.
Local Phone Book: IP phone is saved locally on the phone; Remote Phone Book: IP
Type
phone can be accessed by URL.
Upload the phone book to be imported. Different manufacturers need to download
Please Select a File
different templates.
Select a pre-configured template to automatically fill in such information as features,
Template
preferences, etc.
Model IP phone model.
Account Information required for the IP phone to register with the SIP server.
Line Key Setting Set the shortcut keys of the IP phone.
Call Waiting: When the phone is in a call and another extension calls it, it will enter
the waiting state if this feature enabled, or it will hang up the call if this feature
disabled.
Auto Answer: The IP phone automatically answers after ringing.
Voicemail: When enabled, press the voicemail button on the phone to automatically
Features send *97.
Local Phone Book: When enabled, the configured phone book is automatically
downloaded to the IP phone's address book.
Key As Send:Todetermineif the IP phone directly calls out when pressing the # key.
Remote Phone Book: You can directly view the phone book on the server by using
this feature.
CODEC Set the codec supported by the IP phone.
Enable Auto Provision:Itcan provide automatic configuration function for IP
telephone. By default it is enabled.
Interface: Select the interface to provide automatic configuration for IP phones. By
Config
default it is LAN.
Auto Edit: When it is enabled, the address and port of Auto Provision will be
automatically modified according to the network and SIP settings.
Language: Set the display language of the IP phone.
Admin Password: Set the login password for the IP phone.
NTP Server: Set the NTP server address of the IP phone.
Preference
Time Display Format: Set the time format for the IP phone.
Date Display Format: Set the date format for the IP phone.
Time Zone: Set the time zone for the IP phone.

Page 52
FIBERME Communications LLC
1749 Old Meadow Rd., Suite 401 www.fiberme.com
McLean, VA 22102, USA [email protected]

Page 53