VOI-9300
SIP IP PBX
User Manual
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8.5 SWITCHBOARD (AUTO ATTENDANT SETTINGS) ............................................................................................................39
8.5.1 Prompt Message ...............................................................................................................................................39
8.5.2 Operator Settings ..............................................................................................................................................39
8.5.3 Auto Attendant ...................................................................................................................................................40
8.6 USERS MANAGEMENT...................................................................................................................................................41
8.6.1 Single User Account Opening
8.6.2 Group Users Account Opening
→
→
Add User.................................................................................................42
Add Users...............................................................................................42
8.6.3 Bindings..............................................................................................................................................................43
8.6.4 Delete User Accounts.......................................................................................................................................44
8.6.5 User Information Update..................................................................................................................................44
8.6.6 Function Settings...............................................................................................................................................45
8.6.7 User List..............................................................................................................................................................47
8.6.8 On-Line User List ..............................................................................................................................................48
8.7 ADVANCED SETTINGS....................................................................................................................................................48
8.7.1 Phone Auto Config............................................................................................................................................48
8.7.2 Queue Settings..................................................................................................................................................50
8.7.3 Voice Conference Rooms ................................................................................................................................51
8.7.4 Network Parameter...........................................................................................................................................54
8.7.5 Caller ID..............................................................................................................................................................55
8.8 CALL DETAIL RECORDS QUERY ....................................................................................................................................56
8.8.1 System Call Records query .............................................................................................................................56
8.8.2 User Call Records.............................................................................................................................................57
8.9 UPGRADE & REBOOT ....................................................................................................................................................58
8.9.1 User information Export & Import ...................................................................................................................58
8.9.2 System Upgrade................................................................................................................................................59
8.9.3 Reboot ................................................................................................................................................................59
9. APPLICATIONS..........................................................................................................................................................60
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1. Introduction
1.1. Overview
The VOI-9300 is an embedded Voice over IP (VoIP) PBX Server with Session Initiation Protocol (SIP)
to provide IP extension phone connections for global virtual office of small-to-medium business (SMB)
companys. Equipped with 4 x FXO ports, Ethernet LAN and WAN ports plus Life Line features,
VOI-9300 integrates the telephony network and the data network into a manageable converged
network to provide an efficient and economical PBX for global long distance voice communications.
VOI-9300 IP PBX works with various IP phones (Desktop, WiFi, Bluetooth, and DECT), VoIP
gateways, and analog telephone adapters (ATA) to route calls among client phones, analog phones,
and PSTN network. Call features such as conferencing, auto attendant, and voicemail can be
seamlessly enabled for all phone devices. In addition, it also provides Internet access to all LAN
devices through VPN NAT router.
VOI-9300 IP PBX provides call control and media relay services to SIP clients, and it performs many
primary functions, such as SIP Registrar, SIP Outbound Proxy with media relay, SIP Gateways (FXO),
SIP PBX for extension calls, Auto Attendant Interactive Voice Response (IVR), and Find-Me
Conferencing.
VOI-9300 IP PBX has a built-in suite of PBX applications for supplemental services. This lowers down
the total cost of a converged network enabled by VOI-9300 IP PBX than building separated
infrastructures for legacy telephony network and data network. In addition, with a web-browsable
interface to the data network configuration and voice service provisioning, VOI-9300 brings the
manageability of both networks together to facilitate administration locally and/or remotely.
Note that VOI-9300 requires an IP address, a subnet mask, and its gateway Router IP address for its
own use to connect to Internet. These three are available from your Internet service provider.
VOI-9300 may enable PPPoE or DHCP features to automatically get an assigned dynamic IP from the
ITSP. Please refer to Web Configurations for detailed information.
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2. Features
The VOI-9300 IP PBX is equipped with RJ45 & RJ11 connectors and is featuring as the following:
ꢀ SIP Server supports 50 user registrations and 20 concurrent calls
ꢀ SIP v1 (RFC2543), v2 (RFC3261) with MD5 authentication (RFC2069 and RFC 2617)
ꢀ RJ45 x 2 for Ethernet WAN and LAN ports + RJ11 x 4 for FXO ports + Life Line FXS port
ꢀ Supports ITU-T G.711a, G.711u, GSM/MS-GSM, G.729A/B, VAD and CNG for Speech Codec
ꢀ Configurations by Web Browser
ꢀ Embedded NAT/DHCP Server
ꢀ PPPoE/DHCP Client for Dynamic IP plus NAT, VPN, DNS, and DDNS Clients
ꢀ Support STUN server and DMZ functions for NAT Traversal
ꢀ Support VPN function
ꢀ Support Call features; Call Forward/Waiting/Transfer/Hold, and Voice Conference Room
ꢀ Support E.164 ENUM Dial Number via SIP server
ꢀ Incoming Call Pickup for Group users
ꢀ Incoming Call Ringing for Group users
ꢀ Number Bonding and Call restrictions.
ꢀ Find Me function
ꢀ Extension Pickup for Attendant
ꢀ Bill Rate Table with Voice Mail
ꢀ Interactive Voice Recording (IVR) Settings by XML
ꢀ Programmable Prompt messages
ꢀ On-Line Subscriber Status
ꢀ Remote Firmware Upgraded by HTTP Web Interface
ꢀ Auto Provision Settings
ꢀ Out-Band DTMF (RFC 2833) / In-Band DTMF / Send DTMF SIP Info
3. Standard Compliance
The VOI-9300 IP PBX supports for the following standards
VoIP Protocols:
IETF RFC3261 and RFC 2543 for SIP
SIP Authentication: IETF RFC2069 and RFC 2617 for MD5
Speech Codec:
ITU-T G.711a, G.711u, GSM/MS-GSM, G.729A/B, VAD and CNG
Echo Cancellation: ITU-T G.165/168
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4. Packing Content
Inside the package you should find:
(1) One VOI-9300 IP PBX
(2) One AC100~240V to 12VDC/1A Power Adaptor
(3) One Cat 5 Ethernet Cable
(4) One User Manual CD
5. LED Indicators & Interface Connectors
LED Indicators
On the front panel of VOI-9300, there are 12 LED indicators as the following table
LED
POWER
ACTIVE
ALARM
VPN
Status
ON
Descriptions
Power is Normal.
ON
IP PBX is in Normal Operation
IP PBX is at alarm status
ON
ON
Virtual Private Network function is ON
PSTN Line 1 is enabled and IDLE
PSTN Line 1 is in use
FXO 1
ON
Flashing
ON
FXO 2
FXO 3
FXO 4
LAN
PSTN Line 2 is enabled and IDLE
PSTN Line 2 is in use
Flashing
ON
PSTN Line 3 is enabled and IDLE
PSTN Line 3 is in use
Flashing
ON
PSTN Line 4 is enabled and IDLE
PSTN Line 4 is in use
Flashing
ON
LAN Port is in connection
Flashing
ON
LAN Ethernet data activity
10/100M
WAN
LAN Ethernet port is in connection at 100Mbps
WAN Port is in connection
ON
Flashing
ON
WAN Ethernet data activity
10/100M
WAN Ethernet port is in connection at 100Mbps
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Interface Connectors
1. DC Power
2. FXO ports
12Volt DC / 1A Power Adaptor with 100~240V AC power input
4 FXO ports are for connection to PSTN lines, and numbered 1, 2,
3 and 4 from left to right.
3. Life Line port
Life Line FXS port connects to an analog telephone. When power
is down, the Life Line will switch to FXO port 1 for PSTN line 1.
4. WAN port
5. LAN port
Connect to a broadband ADSL/Cable modem or a WAN router.
Connection to PC for Web configurations or Laptop, IP Phones, or
VoIP Gateways/ATA, etc.
6. USB port
Connect to an external USB drive for backup internal system
storage. Click the Backup icon in Web configurations and follow
instructions to insert the USB connector of an external USB drive.
Note: If PC is directly connected to the LAN port of VOI-9300 for web configurations, please
use the enclosed Ethernet CAT5 Cable.
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6. Installations
WAN
LAN
FXO
Ethernet Switch Hub
ADSL
modem
PSTN
FXS
IP Phones
Internet
Life Line POTS
7. Reset to Factory Default
IP PBX VOI-9300 can be reset back to factory default when IP address is not accessible for web
configurations. The procedures are as follows:
1. Power off.
2. Press the RESET button and hold continuously, then power on.
3. Hold the RESET button until all the LED indicators start flashing for three times. It may take 15-20
seconds to reset, and the RESET button can then be released after flashing.
4. The LED indicators will begin flashing for 3 times again to activate the IP PBX. Note that the
firmware version and IP settings will be reset back to factory defaults and users data base will be
cleared.
5. It is suggested that the user data base be exported and backed up to PC before reset.
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8. IP PBX Configurations by Web Browser
You may enter the IP address from PC Web browser to configure VOI-9300. For example, enter
auto-MDIX for LAN port. If a notebook PC is directly connected to the LAN port of VOI-9300 for web
configurations, the user may use the Ethernet CAT5 cable included in the accessory.
The following Web page shall be displayed on PC. If you have difficulties accessing the
Web page from the PC Web browser, the subnet IP of PC might be different from
192.168.1.xxx. In this case, please refer to Chapter 9 for trouble shooting.
2). Please enter the username and password into the blank field. The default settings are:
Username: admin
Password: 123456
3). Click the “Login” button to enter the VOI-9300 for web configurations.
Whenever you change the setting in each Web page, remember to click the “Submit” button
to save into the non-volatile memory and click the “Reboot” button to activate the new
settings.
WAN & LAN Network IP Address and Mask
NIC
WAN
LAN
IP Address
192.168.139.3
192.168.1.1
Mask
255.255.255.0
255.255.255.0
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4). VOI-9300 provides 100 SIP user ID number accounts which can be configured as well by
Web browser. The preset user ID numbers are from 2001~2010 with same password
123456. The SIP service port is default at 5060.
Resource Status
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8.1 Network
VOI-9300 IP-PBX provides two RJ45 connectors for LAN and WAN ports at 10/100M Ethernet
interfaces. The Network will display the current status for LAN, WAN, DHCP, DDNS, and VPN
settings.
8.1.1 Network Status
Network Status shows all the IP addresses for LAN, WAN, VPN server and VPN clients.
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8.1.2 LAN Setting
LAN Port can be used for IP-PBX to connect to a Notebook PC for configurations. The embedded
DHCP Server will automatically assign IP address through the LAN port.
MAC Address should be unique in the same network, and IP Address must be in the format of
xxx.xxx.xxx.xxx and xxx is from 0 to 255, e.g. 192.168.1.1. Subnet Mask is used for network
segmentation. Please make sure the mask is correct and all the VoIP devices are within the same
network as VOI-9300.
8.1.3 WAN Settings
WAN port is to connect to ADSL modem for Internet access. There are 3 options for WAN settings;
DHCP, Static IP, and PPPoE. The following example shows a Static IP type for WAN Setting.
WAN Link Types
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Static IP mode
Dynamic IP Mode
PPPoE Mode
When VOI-9300 IP-PBX connects to ADSL Modem with PPPOE link, you may need to enter the
account name and password for PPPOE. In addition, you may select the DNS server and enter the
IP address for the First and Second DNS servers.
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8.1.4 DHCP Server
The embedded DHCP server in NAT will automatically assign IP address to the network devices.
DHCP Server Status: To show the current DHCP server status
DHCP Server Start/Stop: To enable/disable DHCP Server
Start/End IP Address: DHCP Server will assign an IP within the start/end IP address range, e.g.
192.168.1.100－192.168.1.200. Note that the start IP and end IP must be
in the same 192.168.1.xxx network.
Mask: Usually 255.255.255.0 for subnet mask
Default Gateway: The IP address for NAT gateway.
DNS Server: The Domain Name Server IP address.
8.1.5 DDNS Settings
DDNS is an abbreviation for Dynamic Domain Name Service. In general, ISP will allocate IP PBX
a dynamic IP address by ADSL PPPoE mode connection when you access Internet, and the IP
address may change day by day. In other words, the IP-PBX IP address may change and can NOT
be reached by a fixed IP address. In such a case, DDNS can be a resolution.
DDNS resolution is used to associate domain name with dynamic IP address acquired from ISP.
Users would only need to know the domain name, and do not care about the current PBX IP address.
www.dyndns.com for free. The IP-PBX supports the DDNS settings for these two most popular DDNS;
3322.org and dyndns.org.
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Service Provider: Choose DDNS server that updates your IP address.
Host Name:
User Name:
Set your host name that need to do DDNS update.
The DDNS server requires you to supply a name and password to update your IP
address. You have to register the user name and password offline to the specific
DDNS server.
Password:
The DDNS server requires you to supply a name and password to update your IP
address. You have to register the user name and password offline to the specific
DDNS server.
DDNS Status:
DDNS Service:
Disable or Enable have to show.
Disable/Enabled DDNS service.
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DDNS Settings
8.1.6 VPN Settings
VPN is namely virtual private network. VPN establishes a temporary and safely connection over
public network (generally Internet). Usually VPN is an extension of an intranet. It helps remote users,
company branches, commercial partners and suppliers to establish a reliable and safely connection
with intranet and guarantees safe data transmission. The IP-PBX supports two kinds of VPN Link:
PPTP VPN server and OpenVPN client.
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VPN Server Configuration for PPTP (Point-to-Point Tunnel Protocol)
VPN Server: You may enable or disable VPN Server Service.
UserName:
Password:
For remote user when connected to VPN server.
For remote user when connected to VPN server.
Press Update to save the configurations, and press Reboot to activate the new configurations.
VPN Client Configuration for OPENVPN
OpenVPN is used when the IP-PBX register as a VPN client to a remote VPN server.
VPN Client Service: You can enable or disable VPN Client Service.
Server Address: The IP address of remote VPN server with format of “address:port”.
Communication: The communication mode between clients and servers.
CA certificate: The CA certificate given by the VPN server.
Client certificate: The Client certificate given by the VPN server.
Client Key: The Client key given by the VPN server.
Press Update to save the configurations, and press Reboot to activate the new configurations.
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8.2 System
The VOI-9300 IP PBX System configurations can be set in this section. The settings include Server
Ports Setting, Rate Setting, DMZ, TrustHost, Music on Hold, Hotlines, Admin Accounts, USB_Disk
Setting, Voicemail Setting, and Time Zone.
8.2.1 SIP Port
The SIP default port number is 5060 for VoIP Applications. The value for SIP port can be from 1 to
65535. Note that the SIP port number must be the same for all the IP-PBX and VoIP phones and
TAs.
HTTP Port : HTTP port number, default at 80.
SIP Port : Default at 5060 .
Encrypt Port : Support SIP packet encryption with VP301/VP306 VoIP Phone and VS110/210/211 TA.
RTP Ports Range : RTP port for VoIP packets. Default at 10000~20000.
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8.2.2 Rate Settings
Rate setting is used to calculate the charges for each call. The IP PBX will generate a call record for
the charges. All the charges are based on this rate table.
The rate is based on the prefix to calculate the charges:
Prefix: 0
Rate: 15 (cent)
Time Unit： 60 (second)
This means when calling a number with 0 prefix (e.g., 01010086)，The rate is 15 cents for every 60
seconds , and the duration less than 60 seconds will be charged the same as 60 seconds. In
addition, the prefix is for the longer one. For example, if there are rates for prefixes 0 and 00, the call
number with 00 prefix will be charged per the rate of 00 instead 0.
If there is no rate for the calling prefix, this implies the rate is 0.
8.2.3 DMZ Settings
DMZ is an abbreviation for demilitarized zone. It is a buffer between safe system and unsafe
system to solve the problem that facilities in the outer net access servers behind firewall in the private
network. The buffer is in a small network zone between enterprise internet and outer net where it can
put some servers that must be public, for example Web server, FTP server and so on.
The IP-PBX supports DMZ by cooperating with routers which provides DMZ function.
The DMZ must be set ON if the DMZ of firewall is activated.
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DMZ mode: You can enable DMZ mode or disable.
External IP Address: External address can be accessed by facilities from Internet.
LAN n: This is the private network segments where the IP-PBX resides in. Update your settings,
Submit -> Reboot.
8.2.4 Trust Host
It is not need to have authentication from the call of Trusted IP Address. If the port is set at 0, all the
ports from this trusted address will not need authentications. The trusted address could be either IP
address or domain name.
The TrustHost address can be added or deleted, and need to be configured in case of the following
conditions;
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(1) Need to work with FXO ports,
(2) Need to connect with VOI-9300,
(3) Need to connect with another SIP Server.
In summary, the TrustHost address can be set whenever the authentication is not needed.
8.2.5 Music ON Hold
The IP PBX will play music when a call is on hold due to the following situations;
(1) When the call is transferred to attendant and waiting for answer.
(2) When the call is hold and waiting for answer.
(3) When the call is transferred and waiting for answer.
You may choose one of the music files for MUSIC ON Hold. You can upload more music , But only
One can Play , The size is unlimited. Propose < 200KB.
Music ON Hold requires a unique file format, and any MP3 files must be converted to .raw format
before upload to VOI-9300 for use of Music ON Hold.
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8.2.6 Hot Lines
Hot line function numbers can be entered into the list. Make sure each of the Hot line function
number is unique, and not the same as any extension number or PSTN numbers.
The Hotline list displays the function number of the PBX, and it will activate the function service when
pressing the corresponding function numbers.
AA: Auto Attendant Voice Prompt Flow Chart
CID Reader: Voice Prompt for the Current IP Phone extension number
Queue Settings Hotline: For Q1-Q4 function settings
Operators Hotline: To assign or unassign extension number as the operator
Exten Settings Hotline:For setting Hotline extension number
My Own Voicemail: To enter Individual voicemail
Voicemail: To enter the wholevoicemail system
Record Prompt: For recording voice prompt
Conference Room1: To enter Conference Room 1
Conference Room2: To enter Conference Room 2
Queue1: Enter Queue1 operators .
Queue2: Enter Queue2 operators .
Queue3: Enter Queue3 operators .
Queue4: Enter Queue4 operators .
IP BroadCast : To Broadcast on the IP Phone exten of Group 0.
Operators : The extension number of operator
Features :
ꢁ
ꢁ
Standard: Standard SIP Protocol
CBCOM: Encryption protocol for SIP and Voice RTP
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Detailed Descriptions for Hotline Function Numbers
Hotline
Functions
Descriptions
Numbers
*8
9
Call pickup
Operator
Press *8# to pickup calls within the same group
Press 9# to call Operator.
Press 112# to enter Auto Attendant IVR followed with the
voice prompts.
112
Auto Attendant (AA)
Any extension phone can join or leave for one of the Queues.
Example: To join as a member of Queue 1 for extension 201
Procedures: From 201 IP Phone,
Dial 1600 -> 1701 -> 123456 (password of Queue1) -> 1 ->
hang up.
1600
Queues Setting
Example: To leave Queue 1 for extension 201
Procedures: From 201 IP Phone,
Dial 1600 -> 1701 -> 123456 (password of Queue1) -> 2 ->
hang up.
Any extension phone can join or leave for the Operators.
Example: To join as one of operators for extension 200
Procedures: From 200 IP Phone,
1601
Operators Setting
Dial 1601 -> 123456 (password of Operator) -> 1 -> hang up.
Example: To quit as one of operators for extension 200
Procedures: From 200 IP Phone,
Dial 1601 -> 123456 (password of Operator) -> 2 -> hang up.
Forward , Find me , Binding, and Password can be set up for
each extension number.
1602
Extension functions
Example: To set extension 202 busy forward to 203
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Procedures: From 202 IP Phone,
Dial 1602 -> 1 (set forward) -> 2 (busy forward)-> 203# -> 1
(submit) -> hang up.
To play & to delete the voicemails for the current IP phone
extension number.
Play Voicemail for
Current Extension
Example: To hear the first voice message and delete
message afterward, and to hear the following voice message
and delete afterward for extension 203.
1603
Procedures: From 203 IP Phone,
Dial 1603 -> 1 -> 7 -> 6 -> 7 -> # to hang up.
To play & to delete the voicemails for the other IP phone
extension number.
Play Voicemail for
Other Extension
Example: To hear the voicemail of extension 203 from
extension 200
1604
Procedures: From 200 IP Phone,
Dial 1604 -> 203 -> 123456(password of extension 203) ->1
-> 7 -> 6 -> 7 -> # to hang up.
1650
1651
1701
1702
1703
1704
Conference Room1
Conference Room2
Queue1
Hotline number of Conference Room1
Hotline number of Conference Room2
Hotline number of Queue1
Queue2
Hotline number of Queue2
Queue3
Hotline number of Queue3
Queue4
Hotline number of Queue4
Hotline Applications
Example:
A customer calls into Operator and wants to talk to the IP extensions 203, but the 203 user are not
near the phone.
Procedures for Operator:
(1) Call Parking for the incoming call
(2) IP Broadcasting for the 203 user
(3) The 203 user may pickup any IP phone and dial 701 to take the waiting call
Example:
If the IP PBX prompts an incoming call with the prompt message “Press 1 for customer service, Press
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2 for technical support, Press 0 for directory”, then you may set as the following;
(1) Call hold 1 for customer service and connect extension 1;
(2) Call hold 2 for technical support and connect extension 2;
(3) Modified the prompt voice messages.
1) Change 1701 hotline number to 1, and 1702 hotline number to 2.
System > Hotlines > Queue1 > Modify of Operation.
Click Submit to finish.
System > Hotlines > Queue2 > Modify of Operation.
Click Submit to finish.
2) Add to the Queues for an existing extension number (208).
Advanced Setting > Queue Setting > Agent Setting
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Submit to add the number 208 for Queue1.
The number will be displayed in the Current Queue Agent List for Queue 1.
Submit to add the number 203 for Queue2.
The number will show in the Current Queue Agent List for Queue2 .
3) Add Voice prompts to Switchboard.
Submit.
It will display voice prompts in the List.
Press the ICON to Activate the Voice.
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The No. 1 of Status will show “1” to indicate activate status.
8.2.7 Admin Account
8.2.8 USB Disk Setting
When you need to store the voice mails, please do the steps as follows;
(1) Insert USB disk
(2) Select “Insert USB Disk” in the current status
(3) Select Yes to enable USB voice mail recording.
(4) Click Submit button.
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When the voice mails is not needed, please do the steps as follows;
(1) Select “Insert USB Disk” in the current status
(2) Select No to disable USB voice mail recording.
(3) Click Submit button.
(4) Pull out the USB disk.
8.2.9 Voicemail Setting
Voicemail is the voice messages of incoming calls. When the call is not answered, the VOI-9300 will
play the voice prompt and the caller may leave voice message. You can play the caller’s voice
message in time. This setting must be compatible with the Voice-Mail Service of Users Function
Setting .
General settings :
MailBox : The e-mail address of mailbox.
User : The user name used by the e-mail address .
Password : The password used by the e-mail address .
SMTP Server : This is the e-mail address used by SMTP of e-mail server domain name .
Use SSL Mode : Need to be checked If SSL mode is used. Do not check if General settings is used.
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SSL for Gmail :
The IP-PBX supports voicemail via Gmail server by using SSL mode.
MailBox: This is used for the sending account in Gmail. This must be registered in Gmail in advance.
User: The user name reistered to Gmail.
Password: The password registered to Gmail.
SMTP Server: Gmail uses port 465 for the SSL SMTP server.
Use SSL Mode: Check to activate the function.
Press Submit to save the configurations, and press Reboot to activate the new configurations
8.2.10 Time-Zone Setting
This provides system date and time information. The time will be recorded for use of CDR, or the CDR
may record wrong timing if the system date and time were not set. The Network Time Service can be
enabled to synchronize with NTP server. You still need to set the TimeZone, Date, and Time. It will
display the synchronized NTP time when you re-enter this setting.
Press Submit to save the configurations, and press Reboot to activate the new configurations.
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8.3 Incoming Calls Settings
The IP PBX provides two ways of incoming calls; one is from local PSTN through the FXO ports, and
the other is from the calls from ITSP. In some case, the ITSP may provide the real PSTN numbers
for the IP PBX from the VoIP.
8.3.1 Calls from FXO ports
The IP PBX provides 4 FXO ports to allow PSTN incoming calls. For any PSTN incoming calls, you
may configure to redirect to either switchboard, extension number, or conference room, etc.
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8.3.2 Calls from VoIP
The IP PBX can function as a terminal CPE to register into a SIP server or another IP PBX. For any
external VoIP calls, for example, 1234 into this IP PBX, you may redirect to either Auto Attendant,
Operator, Conference rooms, etc.
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Registry Address : The SIP Registration Server IP/URL:Port number provided from ITSP, other
IP PBX, or SIP Server.
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User Name : The user name provided from ITSP, other IP PBX, or SIP Server.
Password : The password provided from ITSP, other IP PBX, or SIP Server.
Redirect To: AA, Operators, Conference rooms, etc.
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The list will display registered users with status after submit.
You may also press the icon to modify the setting.
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You may press the icon to delete this setting.
8.3.3 Access Number
Access Number is used to supplement the function of incoming calls from VoIP SIP Server in the
previous section. For some ITSP SIP Servers, one way voice connection may be missing due to
protocol incompatibility. In this case, the Access Number can be set up to register into ITSP SIP
Server and to redirect VoIP calls from ITSP into the IP PBX.
When your registered, for example, 636346 in some SIP server with one-way voice connection, you
may resolve the problem by the following configurations;
(1) In Section 8.3.2 Calls from VoIP
Username: Enter 636346
Redirect to: Enter 636346
(2) In Section 8.3.2 Access Number
Access Number: Enter 636346
Redirect to: Enter AA, Operator, or any Holine function call.
Any VoIP calling at 636346 from ITSP will be redirected into the function calls of VOI-9300, such as
Auto Attendant, Operators, or Hotlines.
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Access Number : The registered number to ITSP or SIP server.
Redirect into : One of the hotline functional calls.
Memo : For memo descriptions.
The Access Number will be displayed after pressing Submit button.
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8.4 Outgoing Call
The IP PBX provides two kinds of outgoing calls; one is via FXO port to local PSTN lines, and the
other is via VoIP calls. The settings are as follows:
8.4.1 Outgoing Calls via FXO ports
The IP PBX is equipped with 4 FXO ports for connection to local PSTN lines. All the extension
number can then make outgoing call to local PSTN numbers. You need to assign the group number
to each of the FXO port to activate the FXO outgoing PSTN line. If any FXO port is not assigned any
group number, the outgoing call for this FXO port will be prohibited and only incoming calls can be
accepted. For outgoing calls, you need to specify the group number for calling PSTN numbers, and
the IP PBX will select one of the available FXO ports from the group to make an outgoing calls. This
make the grouping become flexible in outgoing calls.
Having set the rules for local outgoing PSTN calls, the authority is used to restrict the call. For
example, for authority >=5, only the user extension number with authority >=5 can make an outgoing
call through the FXO ports in this group.
The user extension prefix is used to indicate the group number for making an outgoing call. The delete
digits is used to delete the group number with additional prefix before the dialing string. The external
use is to allow the external user to use the same rule for redial out to local PSTN numbers, when
calling into the IP PBX.
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Here, you can configure outgoing call rules for FXO ports.
Outgoing Desc.: Brief description of this rule.
Authority: Used to restrict user’s call, only user whose Authority is equal or greater than this value can use
this rule to make an outgoing call.
Group NO.: Used to specify the FXO group, for each outgoing call, the IP-PBX selects an available FXO
port from this group.
Dial Prefix: The prefix number when user uses this rule to make an outgoing call.
Dial Strip Bit: Indicates how many bits will be stripped from the dialed number.
ppend Prefix: Indicates what added on the dialed number.
Extern User Control: If select “Enable”, users from the third party system can use this rule to make an
outgoing call.
Example:
When one user press 056789 to apply the outgoing rule, the IP PBX will use one of the FXO port in the
group 1, delete 0 (one digit), The final dialing string will be 56789 to the PSTN line.
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8.4.2 Outgoing Calls via VoIP
SIP Trunk is a logical pipeline connecting IP-PBX and SIP ITSP. By SIP Trunk, extension users in this
system can communicate with users in the third-party VoIP system.
The VOI-9300 IP-PBX supports outgoing call through SIP Trunk, connecting with the third-party VoIP
system, which means, we can either call users in other system, or call PSTN phone through other system.
There are two ways in connecting with the third party system:
1) The third party system provides an account and a password; we use this account and a password to
authenticate outgoing call. In this case, we need to fill “UserName” and “Password” fields when we add an
outgoing rule via VoIP.
2) The third party system set us as a trust host, do not authenticate our outgoing call but accept it
directly. In this case, no need to fill “UserName” and “Password”.
When using the first way, the caller number of our outgoing call is the account the other system provides.
When using the second way, the caller number of our outgoing call is the transformed number of our
extension number. The transformation depends on the values of “User Prifix” and “User Strip Bit” fields.
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Description: Brief description of this rule.
Authority: Used to restrict user’s call, only user whose Authority is equal or greater than this value can use
this rule to make an outgoing call.
IP/Domain Address: The IP address of the third party system.
UserName: The account provided by the third party system.
Password: Password corresponding to the account.
Dial Prefix: The prefix number when user uses this rule to make an outgoing call.
Dial Strip Bit: Indicates how many bits will be stripped from the dialed number.
Append Prefix: Indicates what added on the dialed number.
User Prefix: Indicates what added on the caller number.
User Strip Bit: Indicates how many bits will be stripped from the caller number.
Feature: If encrypt SIP signaling or not.
Outside User Limit: If select “Enable”, users from the third party system can use this rule to make an
outgoing call.
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8.5 SwitchBoard (Auto Attendant Settings)
The IP PBX provides an Auto Attendant for user to call. When receiving an incoming call, the auto
attendant will play a welcome prompt message or call another extension number. The settings
include welcome prompt message, operator, and auto attend.
8.5.1 Prompt Message
You may choose the prompt message for the auto attendant to play while receiving incoming calls.
You need to upload the messages to the prompt list for setting choices.
Note: The prompt file must be GSM format.
8.5.2 Operator Settings
When a transfer call can not be transferred to the desired extension number, the call will be
transferred to operator. The operator can be any extension or PSTN number with assigned priority,
and the call will be forwarded by priority.
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8.5.3 Auto Attendant
Auto Attendant will handle all the incoming calls when no one can answer in the company.
There are two ways for attendant. The first one is Prompt with playing answering prompt messages,
and the second one is Phone with direct transfer to the phone number.
ꢁ
Prompt : An answering prompt message will be played to answer the incoming call. Make
sure the answering message is uploaded and chosen. The IP PBX provides many options for
time durations. If the time durations are not set, the handling incoming call will be always
effective.
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Phone : All the incoming calls will be transferred to the preset phone number. The phone
number can be either extension number or PSTN number.
ꢁ
ꢁ
Attend Way : You can select the attend way is prompt or phone.
Promot File : If you selected the prompt way . You can choose one prompt file to be use .
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ꢁ
ꢁ
Number : If you select the Number way . The extension number will answer the call.
Time/Week/Date/Month: The time of these time field set (Time & Week & Date & Month), the
extension user will answer the call.
8.6 Users Management
This section describes the account opening, closing, and management. The extension number (also
known as user name) must not exceed 32 digits. For easy management, it is recommended to assign
different prefix number for different departments.
A user number should have the following;
(1) User name: The user name is a string of number digits with length less or equal to 32. For
easy management, it is recommended to assign different prefix for different departments.
(2) Password: Every user name will have its own password with length less than or equal to 32
alpha-numeric digits. The CPE VoIP phone must use the same password for authentications.
(3) Call Authority: from 0-15. This is used to restrict the call authority. The user authority must
be equal or greater than the predefined authority to make the defined outgoing calls.
(4) User Group: Each user name can belong to one or multi-groups. User can answer
incoming calls for another user within the same group by pressing “*8”. There are 16 groups can
be assigned.
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8.6.1 Single User Account Opening → Add User
You may add single user account in this section by entering user name, password, groups, and call
authority. Remember the length should not exceed 32.
Note: when the number of registered users reach the limit, the function of add user will not work.
8.6.2 Group Users Account Opening→ Add Users
You may create group users accounts and assign group numbers with priority to all the accounts.
You may also assign password to each account or generated by IP PBX. After settings, you may
display all the created accounts.
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8.6.3 Bindings
Extension binding is used to bind many extension numbers together as a group. When calling to one
of the extension, the other binding extension will ring as well. This is also referred to as Group
Ringing.
(1) User-ID: To enter extension number.
(2) Bind-ID: To enter the binding ID. The extension number with same binding ID will belong to the
same binding group.
(3) The binding list will display the extension numbers.
Note that one extension number can belong to multi-binding groups.
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8.6.4 Delete User Accounts
There are two ways for user account deletion; one is for single account deletion, and the other is for
group account deletions. For single account deletion, you need only to enter the extension number.
For group account deletions, you need to enter a range of extension numbers. The IP PBX will
delete all the user information for the extension within the range. Note that the information will not be
recovered once deleted.
8.6.5 User Information Update
In this section, you may modify password, name, department, authority and group.
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ꢁ
ꢁ
Inquiry : Choose one User-ID or User Name for search .
Condition (Fuzzy Query) : Enter the number or user name for search.
This page will show the Modify User settings.
8.6.6 Function Settings
There are three functions provided as follows;
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ꢁ
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Inquiry : Choose one User-ID or User Name for search .
Condition(Fuzzy Query) : Enter the number or user name for search.
This page shows the Modify User's Function setting.
(1) Call Forward
ꢀ Unconditional Forward: Any incoming call to this extension will be forwarded directly to the set
extension number.
ꢀ No-Response Forward: When no answer in one minute, the call will be transferred to the set
extension number.
ꢀ Busy Forward: When busy, the new incoming calls will be transferred to the set extension number.
ꢀ Off-Line Forward: When the extension number is off-line (unregistered), all the incoming call will
be transferred directly to the set extension number.
(2) Find Me
When one user is set for “Find Me”, the call will be transfer to the first extension number when no
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answer for the incoming call. If no answer again, the call will be transferred to the next extension
number until the call is answered. The maximum is 6 extensions numbers. After that, the call
will be disconnected.
(3) Telephone Binding
The telephone binding is to bind the extension number with a PSTN number. For incoming call
to the extension number, the PSTN phone number will ring simultaneously.
(4) Voice Mail
You may enable the voice mail for extension numbers.
You may to choose one for Approach to store the Voice-mail method .
1) Send msg to mailbox, not saved on server : It will only send the message to the Email
Address without saving a copy on the IP PBX.
2) Send msg to mailbox, and saved on server : It will send the message to Email address
with a copy saved on the IP PBX.
Note :
(1) Every user may select only one at a time out of the three functions.
(2) You need to check on the icon to select the desired function.
(3) The dialing number must follow the rules for outgoing calls.
8.6.7 User List
There are two kinds of list; one for all the users, and the other for the on-line registered users.
The list will show the current setting status for each user as follows;
(1) Normal
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(2) Unconditional Transfer
(3) Call Transfer; including busy, off-line, and no answer transfer.
(4) Find Me
(5) Telephone Binding
You may modify and update all the user settings directly from the entry of the list.
8.6.8 On-Line User List
The on-line users are for the current registered users.
8.7 Advanced Settings
The advanced Settings cover some call waiting, voice conference rooms, and IVR upload process.
8.7.1 Phone Auto Config
The IP-PBX system can configure automatically the IP Phone with auto-configuration capability by HTTP
protocol.
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Firstly, you need to set up auto configuration for the IP Phone. The following example is for reference
only.
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Auto Configuration: Select the HTTP protocol.
HTTP Server: IP address or URL of the IP-PBX.
HTTP File Path: Configuration file’s saved path, it must be “/conf/”.
Secondly, you need to configure for the IP PBX.
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User-ID : The extension number of the telephone.
Server Address : The IP address or URL & port of the IP PBX.
Phone's Mac Address : The Mac address of the IP Phone.
ꢁ
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8.7.2 Queue Settings
Call waiting is to queue all the incoming calls and to assign based on rules to the desired extension
numbers. If all the available extensions are busy, the system will play the waiting music for the
queuing incoming calls. Once available, the system will connect to the desired extensions.
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The IP PBX provides 4 call waiting queues. The waiting queues can be set to a simple calling
center, and the user may define individual waiting queues for its own purpose.
The waiting queues can be configured by the IE Web browser to define the functions as follows;
(1) Update new incoming calls;
(2) Setting the password
(3) Set the desired extension
(4) Set the waiting time length
(5) Add/Delete waiting number
Functions:
1） Update the call waiting list
2） Modify the call waiting information
The call waiting password will be required when the call is entering the waiting queue.
For each waiting call, the IP PBX may provide one of the following call assignment:
(1) Round-Robin: for any incoming call, the call will be forwarded to the next extension
number.
(2) Random: The incoming call will be randomly assigned to the extension numbers.
(3) The least-answer: the new incoming call will be transferred to the extension with the least
answering.
(4) The longest idle: the new incoming call will be transferred to the extension with the longest
idle time.
(5) All Ringing: the new incoming call will ring all the extension numbers. Anyone pick up the
phone will answer the call. Waiting length means the maximum numbers for call waiting.
3） To Add/Delete the waiting list
The user may add/delete the call waiting queues.
8.7.3 Voice Conference Rooms
The IP PBX provides two standard voice conference rooms. The details are as follow;
(1) Conference Room Password
When a user calls to the conference room, he will be required to enter the password, or the call
will be denied. It is recommended to set the password for the conference room.
Please refer to the conference room setting for password.
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(2) Maximum attendants of conference room
If set at 0, it has no limitation. Please refer to the conference room settings.
(3) Web Control of Conference Room
There are three icons for the three operation functions;
(1) The extension numbers are in conversation dialog status
(2) The extension is in monitoring status.
(3) Kick out the extension number. The extension will be forced out in 5 minutes.)
(4) IVR Control of Conference Room
The user after entering the conference room may press * key for an IVR playback. The user may
interact with the IVR messages.
General User Extension:
1) Make oneself mute. The user voice will not be heard in the conference room. To let one’s voice
heard in the conference, just repeat the same procedure as mute. Press * key for an IVR
playback, then press 1 key to resume conference attending.
2) Make oneself manager. The user needs to enter the manager password. Please refer to
conference room settings for the manager password.
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Manager has the following functions:
1, Make oneself mute. The user voice will not be heard in the conference room.
2. Make the current conference room in lock, and any user trying to enter the conference room will
hear the prompt message “Conference Room is in Lock”, and get disconnected. To unlock the
conference room, simply repeat the same procedure in lock. That means to press * first, and then
press 2 to unlock the current conference room.
3. To kick out all the general users. (Note that all managers will not be kicked out.).
4. To mute all the general users. All the general users in the conference room will not be able to
speak but to listen only.
5. To disable mute all the general users. All the general users in the conference room will be able to
speak and listen. (Note that if the user mutes oneself he needs to disable mute by himself to
speak in the conference room. The manager can not disable mute for the user.)
6. Conference Invitation.
If you want to invite an outside user to join the conference room, you may follow the step by the
IVR message operations. This outside user can be registered in the IP PBX or the PSTN number.
Note: Conference invitation is making an invitation only and do not acknowledge if invitation is
successful or not. If not successful, the invitation will repeat after 60 seconds. If not successful
after three times of conference invitation, the invitation will stop without any notice.
(5) Support IVR Conference Invitation
You may make conference invitation by IVR. Please refer to the IVR conference room control.
(6) Modify Voice conference room service number
You may change the conference room service number if the default number is not desired.
The conference room settings are as in the following page.
The conference room password is used to enter the conference room. The maximum capacity is for
the maximum number of users in the room.
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8.7.4 Network Parameter
The network parameters allow to enable/disable IP network functions.
Support Voice Priority Tag (TOS/DSCP)
TOS (Type of Service) and DSCP (Differentiated Services Code Point) are used to set different priority
to data packet and data flow, thus enabling QoS in IP communication.
TOS has four values; they are minimum delay (TOS_LOWDELAY), max throughput
(TOS_THROUGHPUT), max reliability (TOS_RELIABILITY) and minimum cost (TOS_MINCOST).
TOS can be one of them in practice.
DSCP field is a superset of TOS, its definition is backward-compatible with TOS, and its value can be
from 0 to 63, with 0 for minimum priority, 63 for max priority.
Support Tag-based VLAN 802.1P/802.1Q.
VLAN 802.1P/802.1Q are for local area network standard recommended by IEEE, which can partition
network users from different physical locations into several logical subnets. The VOI-9300 IP-PBX
supports this standard, and can add special VLAN Tag to data frames passing through.
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8.7.5 Caller ID
This allow to configure the caller ID fuctions and to show the incoming call numbers.
Default is “Caller ID after 1st Ring (FSK)”.
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8.8 Call Detail Records Query
The IP PBX keeps call records for 2 months. You may have two ways for record queries. One is to
query for the whole IP PBX, and the other is for the specific user. If you want to keep the call records,
you may copy to USB disk before erased.
8.8.1 System Call Records query
You may query all the calls during the specified time frame and export to local storages.
By Month
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By Time
8.8.2 User Call Records
You may query all the calls for certain user during the specified time frame and export to local
storages.
Example :
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8.9 Upgrade & Reboot
8.9.1 User information Export & Import
Before upgrading the VOI-9300 IP-PBX or reset to factory defaults, you may export all the user
information to local PC storage, and import back after the upgrade or default settings are done.
Export: Click on “DOWNLOAD” button and select “Save as New File”.
Import: Select the desired file, and upload.
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8.9.2 System Upgrade
VOI-9300 IP-PBX provides WEB upgrading. Before upgrading, make sure the following;
(1) Get the desired upgrade version. The VOI-9300 support two different ways for upgrade; Web
GUI Upgrade and SIP Server Upgrade.
(2) Backup a copy of user information by import/export data files.
(3) Make sure the power is on while upgrading.
Then you may upgrade per the web instruction procedures.
8.9.3 Reboot
Reboot is used to enable the changes. Note that for Factory Default Setting, please refer to the
chapter. 7 Reset to Factory Default.
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9. Applications
Applications of IP PBX under Firewall with DMZ
This will protect corporate network security while allowing VOI-9300 to work as IP-PBX for VoIP
applications. When VOI-9300 IP-PBX is operating under the corporate firewall, remember to
enable and open the following service port numbers for VoIP applications.
ꢀ
ꢀ
TCP Port：22, 80, 1723 for Telnet, Http, and PPTP.
UDP Port：5060, 1194, 10000-20000 for SIP, OPENVPN, and Voice RTP Range.
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Applications of IP PBX with ADSL
This VOI-9300 supports PPPOE to work with ADSL and to integrate IP-PBX into the corporate
network for VoIP applications.
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