Teletronics IP-PBX User Manual

EZLoop® IP-PBX Enterprise SIP Server

2 Choke Cherry Road, Rockville, MD 20850

sales@teletronics.com

www.teletronics.com

CH1. Overview..................................................................................................4

1.1 Specifications ...........................................................................................................4

1.2 Hardware Overview ..................................................................................................6

1.2.1 Front Panel and LED Indication ...........................................................................6

1.2.2 Back Panel...........................................................................................................7

CH2. Start to configure EZLoop® IP-PBX Enterprise SIP Server ...............8

2.1 Step 1.........................................................................................................................8

2.2 Step 2.........................................................................................................................8

2.3 Step 3.........................................................................................................................8

2.4 Step 4 ........................................................................................................................8

2.5 Step 5: To configure basic and essential configurations......................................9

2.5.1 Network Configuration..........................................................................................9

2.5.2 Extension Configuration.....................................................................................10

2.5.3 Trunk Configuration............................................................................................12

CH3. Full Web Configurations..................................................................... 17

3.1 Configuration ..........................................................................................................18

3.1.1 IP PBX ...............................................................................................................18

3.1.2 Extension ...........................................................................................................21

3.1.3 Trunk..................................................................................................................24

3.1.4 SIP Trunk Reg....................................................................................................28

3.1.5 Routing Table .....................................................................................................30

3.1.5.2 Incoming Call Rule..........................................................................................33

3.1.6 Dial Group..........................................................................................................34

3.1.7 Speed Dial .........................................................................................................35

3.1.8 Other Setting......................................................................................................37

3.2 Information..............................................................................................................39

3.2.1 Subscriber..........................................................................................................39

3.3 Management............................................................................................................40

3.3.1 Network..............................................................................................................40

3.3.2 TimeZone...........................................................................................................41

3.3.3 SMTP Setting.....................................................................................................42

3.3.4 User Account......................................................................................................42

3.3.5 Firmware Upload................................................................................................43

3.3.6 Music Upload .....................................................................................................44

3.3.7 Import Setting.....................................................................................................44

3.3.8 Export Setting ....................................................................................................45

3.3.9 Flash Clean........................................................................................................46

3.4 Reboot System .......................................................................................................47

CH4. Application Setting.............................................................................. 48

4.2 Customize Ring Back Tone (Transferring Tone) .................................................56

4.3 Call Features ...........................................................................................................56

4.3.1 Authentication ...................................................................................................57

4.3.2 Automated Attendant..........................................................................................57

4.3.3 Call Transfer.......................................................................................................57

4.3.4 Blind Transfer (Client based)..............................................................................57

4.3.5 Call Forward on Busy (Client based)..................................................................57

4.3.6 Call Forward on No Answer (Client based) ........................................................58

4.3.7 Call Forward Unconditional (Client based).........................................................58

4.3.8 Call Hold/Retrieval (Client based) ......................................................................58

4.3.9 Call Routing .......................................................................................................58

4.3.10 Call Waiting (Client based)...............................................................................58

4.3.11 Caller ID ...........................................................................................................58

4.3.12 Do Not Disturb (Client based) ..........................................................................58

4.3.13 Flexible Extension Logic ..................................................................................58

4.3.14 Music On Hold ................................................................................................58

4.3.15 Music On Transfer ...........................................................................................58

4.3.16 Call Pickup (Global Call Pickup) ......................................................................58

4.3.17 Three-way Conference (LP388).......................................................................58

4.3.18 Time and Date .................................................................................................58

4.3.19 Trunking (EZLoop® PSTN Gateway FX04/06).................................................58

4.3.20 VoIP Gateways (EZLoop® PSTN Gateway FX04/06; )....................................58

4.3.21 Voice Mail to e-mail..........................................................................................59

CH5. Appendix-Application between CPE device and EZLoop® IP-PBX

Enterprise SIP Server. .................................................................................. 60

5.1 Extensions register to EZLoop® IP-PBX Enterprise SIP Server with number

101 to 106. All Extensions can talk to each other. .....................................................61

5.2 The call will be forward to MailBox if the extension 101 is busy or no answer.65

5.3 The Trunk (EZLoop® PSTN Gateway FX04/06) can also register to EZLoop® IP-

PBX Enterprise SIP Server (registered number 888).................................................67

5.4 EZLoop® IP-PBX Enterprise SIP Server can register to ITSP as a SIP-Trunk. ..71

5.5 All of the Extensions can call out to local PSTN via EZLoop® PSTN Gateway

FX04/06..........................................................................................................................73

5.6 All of the Extensions can call out to Mobile Phone via ITSP...............................74

5.7 User in PSTN side should be able to contact with Extensions via EZLoop®

PSTN Gateway FX04/06................................................................................................78

5.8 User in ITSP side should be able to contact with Extensions ............................79

5.9 User in ITSP side can call out to local PSTN via EZLoop® PSTN Gateway

FX04/06..........................................................................................................................81

5.10 Traveler can call back to EXT, and Traveler can also call to local PSTN and

Mobile phone number...................................................................................................83

CH1. Overview

The EZLoop® IP-PBX Enterprise SIP Server is the next generation all-in-one IP PBX system for small to medium enterprise. It is also designed to operate on a variety of VoIP applications, such as auto-attendant, voice conference, call transfer, call pick up and IPbased communications. With the tiny box, small to medium enterprise or homes can use it to access the Internet and to make VoIP phone calls. Customers can select different suite and optional products to meet their request. It can be integrated with EZLoop® PSTN Gateway FX04/06 and can provide PSTN access function; it can also provide extensions. With flexible and full functionality, EZLoop® IPPBX Enterprise SIP Server can give a complete transition from traditional PABX to the new generation IP-PBX.

1.1 Specifications

 Protocol

 SIP (Session Initiation Protocol)  Call Features

 Authentication

 Automated Attendant

 Call Transfer (CPE based)

 Blind Transfer (CPE based)

 Call Forward on Busy (CPE based)

 Call Forward on No Answer (CPE based)

 Call Forward Unconditional (CPE based)

 Call Hold/Retrieval (CPE based)

 Call Routing

 Call Waiting (CPE based)

 Caller ID

 Do Not Disturb (CPE based)

 Flexible Extension Logic

 Music On Hold

 Music On Transfer

 Call Pickup

 Three-way Conference ()

 Time and Date

 Trunking ( EZLoop® PSTN Gateway FX04/06)

 VoIP Gateways ( EZLoop® PSTN Gateway FX04/06)

 Voice Mail to e-mail

 Call Detail Records

 Codecs

 G.711 (A-Law & _-Law)

 G.729  Technical Features

 Subscriber NAT transversal

 Phone set record Greeting

 Management: Web Browser Management

 HTTP upgrade firmware and ring back tone file

 Export/Import configuration

 Network Interface: 1WAN 1LAN

 DTMF: in-band, RFC2833, SIP-Info

 Network: Support Fixed IP, DHCP, and PPPoE mode  Capacity

 100 register users

 20 concurrent calls  Dimension

 17.5 x 12.5 x 3.2 cm

1.2 Hardware Overview

1.2.1 Front Panel and LED Indication

 Power: Light on when EZLoop® IP-PBX Enterprise SIP Server is powered on.  Status: Light on when system is ready.  Alarm: Light Flash when system is upgrading software, please do not unplug power

when Alarm is flashing.  SIP Trunk: Light on when EZLoop® IP-PBX Enterprise SIP Server successfully

registered to all of the enabled SIP Trunks; Light flash when EZLoop® IP-PBX

Enterprise SIP Server failed to register to one of the enabled SIP Trunks; light off

when there is no SIP Trunk has enabled.  CDR: EZLoop® IP-PBX Enterprise SIP Server can output Call Detail Records to

external computer. User has to execute CDR program on computer, when EZLoop®

IP-PBX Enterprise SIP Server is ready to connect with CDR server and output data,

this indication will light on.

Note:

• CDR Function can only work in local area network. Please prepare the CDR server under LAN.

• The CDR server is proprietary, for more information about CDR, please contact with the sales person of .

 NET: Display Network status. If WAN port of EZLoop® IP-PBX Enterprise SIP Server

is under Fixed IP mode, LCD will light on. If WAN port is under DHCP or PPPoE mode, and EZLoop® IP-PBX Enterprise SIP Server succeeds in getting IP, LED will be flashing. If WAN port is under DHCP or PPPoE mode, and EZLoop® IP-PBX Enterprise SIP Server fails to get IP, LED will light off.

 WAN

• LINK/ACT: Light on when WAN port is connected to network. Flash when data is transmitting or receiving.

• 10/100: Light on when network rate is 100 Mb/s, and light off when network rate is 10 Mb/s.

 LAN

• LINK/ACT: Light on when LAN port is connected to network. Flash when data is transmitting or receiving.

• 10/100: Light on when network rate is 100 Mb/s, and light off when network rate is 10 Mb/s.

1.2.2 Back Panel

 Reset: Network and Login information will return to default values.  LAN/WAN: RJ-45 socket, complied with Ethernet 10/100base-T.

The pin-out is as following:

PIN 1, 2: Transmit

PIN 3, 6: Receive

 12V DC: Input AC 100V~240V;output DC12V

CH2. Start to configure EZLoop® IP-PBX Enterprise SIP Server

2.1 Step 1

Connect LAN port of EZLoop® IP-PBX Enterprise SIP Server with PC via crossover cable or connect with Switch/ Hub via straight through cable.

2.2 Step 2

Prepare one computer, and change the IP address to be 192.168.123.12x with subnet mask 255.255.255.0.

2.3 Step 3

Open browser and link to default LAN IP address of EZLoop® IP-PBX Enterprise SIP

Server “192.168.123.123” with default port number 10087, i.e.

http://192.168.123.123:10087

2.4 Step 4

Login EZLoop® IP-PBX Enterprise SIP Server with default userID: “root”, and no password. After login EZLoop® IP-PBX Enterprise SIP Server, user can start to

configure basic and essential configurations.

2.5 Step 5: To configure basic and essential configurations

To make EZLoop® IP-PBX Enterprise SIP Server work have to set some basic and essential configurations, those include Network, Extension (FXS and IP Phone devices), and Trunk (FXO devices).

2.5.1 Network Configuration

Enter Management  Network to configure WAN and LAN IP.

 WAN

• Mode: Select EZLoop® IP-PBX Enterprise SIP Server WAN port network mode to be Fixed IP, DHCP or PPPoE.

• IP Address/NETMASK/GATEWAY: If user has set EZLoop® IP-PBX Enterprise SIP Server to be fixed IP mode. User need to input IP address/Subnet Mask/ Default Gateway.

• DNS: Input DNS address.

• PPPoE ID: If user select PPPoE mode, here can input PPPoE account ID.

• PPPoE PWD: If user select PPPoE mode, here can input PPPoE account

password.

• Mac: Mac address of EZLoop® IP-PBX Enterprise SIP Server WAN port. The Mac address cannot be modified.

 LAN

• IP Address: Input IP address for LAN port of EZLoop® IP-PBX Enterprise SIP Server.

• NETMASK: Input Subnet Mask for LAN port of EZLoop® IP-PBX Enterprise SIP Server.

• Mac: Mac address of EZLoop® IP-PBX Enterprise SIP Server LAN port. The Mac address cannot be modified.

Press Apply to save configuration, or press Cancel to quit configuration.

2.5.2 Extension Configuration

User has to set Extension account for other device to register on EZLoop® IP-PBX

Enterprise SIP Server.

Enter Configuration  Extension to configure Extension data. On screen will show 100 sets Extension. User can press Modify to add new Extension or modify configured Extension data. Press Delete will delete the specified Extension.

After press Modify can input detail setting for Extension.

 Extension Number: Assign the number of Extension. This number is also the register

name for device.  Password: Assign the register password for device to register on EZLoop® IP-PBX

Enterprise SIP Server.  Call Group: You can use the Call Group parameter to assign an Extension to one or

more groups.  Pickup Group: You can use the Pickup Group option in conjunction with this

parameter to allow a ringing phone to be answered from another extension.

Note:

 The Pickup Group option is used to control which Call Groups a channel may pick

up—a channel is given authority to answer another ringing channel if it is assigned to the same Pickup Group as the ringing channel’s Call Group. By default, remote ringing extensions can be answered with *8.

 You can define multiple Call Groups and Pickup Groups for one Extension by a

“comma”. For example, you can input “1,3,5” into Call Group or Pickup Group.

 DialPlan: Define the dialing plan for Extension. It specifies the location of the

instruction used to control what the phone is allowed to do, and what to do with

incoming calls for this extension. In this field, you can Choose 5 dial level for

Extension, including [ext-only], [ext+R1], [ext+R12], [ext+R123], [ext+allroutes]. You

can define an “Outgoing call” record, to a certain Route Level, as R1, R2…, etc. [ext-

only] means this subscriber can only call to Extension. [ext+R1] means the subscriber

with such DialPlan can call to Extension and Route Level with R1. [ext+R12] means

the subscriber with such DialPlan can call to Extension and Route Level with R1 and

R2. [ext+R123] means the subscriber with such DialPlan can call to Extension and

Route Level with R1, R2 and R3. [ext+allroutes] means the subscriber with such

DialPlan can call to Extension and Route Level with R1, R2, R3 and R4.

Note:

• For more information about Route Level, please refer to the user manual: 3.1.5.1

Outgoing Call Rule.

 Keypad: User can select Keypad type to be RFC2833, In-band, or SIP-Info. The

setting should be also match the Keypad setting of Extension device.  NAT Traversal: If the Trunk device is behind a device performing NAT, such as

firewall or router, and need to register to EZLoop® IP-PBX Enterprise SIP Server on

public network, then user has to enable this function. Enable NAT Traversal to force

EZLoop® IP-PBX Enterprise SIP Server to ignore the contact information for the

Extension and use the address from which the packets are being received.  RTP Mode: User can choose for two type of RTP mode, one is Routed Mode another

is Direct Mode. The voice media will be routed “Peer-to-Peer” if two clients are both

setting to Direct Mode. This way will improve the voice quality and reduce the

performance wastage of the EZLoop® IP-PBX Enterprise SIP Server.

Note:

• If one peer set to Direct Mode but another peer set to Routed Mode, the result will become to Routed Mode.

• Voice media will still go through the EZLoop® IP-PBX Enterprise SIP Server if the EZLoop® IP-PBX Enterprise SIP Server needs to detect DTMF.

• If you enable the NAT Traversal function for Extension, the RTP mode will change to Routed Mode directly; this way will avoid the “one-way voice” or “no voice issue” of VoIP.

• If the both peers are under different subnet, or one peer is under Public IP but another one is under Private IP, we strongly suggest you to set the RTP

mode to Routed Mode to avoid some unexpected voice problems.

 Mail Box: User can select to disable or enable mail box function. If this function is

enabled, user has to input e-mail address for the Extension. When having voice mail of incoming call, system will send this voice mail to the specified e-mail address.

Note:

• Please remember set the SMTP in the page of Management  SMTP Setting to activate the Voice Mail to E-mail.

• If the EZLoop® IP-PBX Enterprise SIP Server got a new message, it will send the message to the user by email immediately.

Press Apply to save configuration, or press Cancel to quit configuration.

Note:

• For more information about Extension setting, please refer to user’s manual

CH3. Full Web Configurations

2.5.3 Trunk Configuration

User has to set Trunk account for Trunk (FXO device, e.g. ) to register to EZLoop® IPPBX Enterprise SIP Server or set some necessary configuration for SIP trunk (For more application, please go to…….). Enter Configuration  Trunk to configure Trunk data.

On screen will show 20 sets Trunks. User can press Modify to add new Trunk or modify configured Trunk data. Press Delete will delete the specified Trunk.

After press Modify can input detail setting for Trunk.

 Trunk Number: Assign the number of Trunk. This number is also the register name

for Trunk device.

Note:

• The Trunk Number can also be a “Trunk ID”. In the Routing Table page, you should define the destination of prefix route. When you define the prefix route, you should set the Trunk ID (Trunk Number) in the Trunk page first; then you could input the correct Trunk ID in the Destination field.

 Password: Assign the register password for device to register on EZLoop® IP-PBX

Enterprise SIP Server.

 Host: Setting the Host to Dynamic will require the trunk to register the EZLoop® IP-

PBX Enterprise SIP Server so that the EZLoop® IP-PBX Enterprise SIP Server knows how to reach the trunk. You can also set the Host to an IP address or FQDN if you set the Host to [Pre-define]. There will be a field called [Address] appear when you choose Host to [Pre-define]. This limits only where you place calls to, as the user is allowed to place calls from anywhere.

 DialPlan: Define the dialing plan for Trunk. It specifies the location of the instruction

used to control what the phone is allowed to do, and what to do with incoming calls for this extension. In this field, you can Choose 6 dial level for Extension, including [frompstn], [ext-only], [ext+R1], [ext+R12], [ext+R123], [ext+allroutes]. You can define an “Outgoing call” record, to a certain route level, as R1, R2…, etc. [from-pstn] is used for Trunk only. [ext-only] means this subscriber can only call to Extension. [ext+R1] means the subscriber with such DialPlan can call to Extension and Route Level with R1. [ext+R12] means the subscriber with such DialPlan can call to Extension and Route Level with R1 and R2. [ext+R123] means the subscriber with such DialPlan can call to Extension and Route Level with R1, R2 and R3. [ext+allroutes] means the subscriber with such DialPlan can call to Extension and Route Level with R1, R2, R3 and R4.

Note:

• For more information about Route Level, please refer to the user manual: 3.1.5.1

Outgoing Call Rule.

 DTMF: User can select DTMF type to be RFC2833, In-band, or SIP-Info.  NAT Traversal: If the Trunk device is behind a device performing NAT, such as

firewall or router, and need to register to EZLoop® IP-PBX Enterprise SIP Server on public network, then user has to enable this function. Enable NAT Traversal to force EZLoop® IP-PBX Enterprise SIP Server to ignore the contact information for the Trunk and use the address from which the packets are being received.

 RTP Mode: User can choose for two type of RTP mode, one is Routed Mode another

is Direct Mode. The voice media will be routed “Peer-to-Peer” if two clients are both setting to Direct Mode. This way will improve the voice quality and reduce the performance wastage of the EZLoop® IP-PBX Enterprise SIP Server.

Note:

• If one peer set to Direct Mode but another peer set to Routed Mode, the result will become to Routed Mode.

• Voice media will still go through the EZLoop® IP-PBX Enterprise SIP Server if the EZLoop® IP-PBX Enterprise SIP Server needs to detect DTMF.

• If you enable the NAT Traversal function for Extension, the RTP mode will change to Routed Mode directly; this way will avoid the “one-way voice” or “no voice issue” of VoIP.

• If the both peers are under different subnet, or one peer is under Public IP but another one is under Private IP, we strongly suggest you to set the RTP

mode to Routed Mode to avoid some unexpected voice problems.

 Port: You can use this to define the SIP signal port if you want to listen on a

nonstandard SIP signal port.

 External Server Address: This field will allow you to set the domain in the SIP From

URI. Setting this will avoid some unexpected issue if the service provider needs this for authentication.

 Maximum Channels: This will limit the maximum channels for this Trunk. For

example, you set 2 into this field; only 2 outgoing calls could go via this Trunk. Default is no limit.

 Outbound Caller ID: Some service provider will require the correct registered caller

ID if it got an incoming call. Default the EZLoop® IP-PBX Enterprise SIP Server will send the Extension’s caller ID to this Trunk, if you set empty here.

Note:

• Normally, SIP From URI will contain the Extension’s calling ID and EZLoop® IP- PBX Enterprise SIP Server’s IP address, but some ITSP may reject this call due to some security issue. You can modify the Calling ID and IP/ Domain in the fields of [External Server Address] and [Outbound Caller ID] when the call is going via the EZLoop® IP-PBX Enterprise SIP Server to the Destination (Trunk) to avoid such security issue.

• If you set a FXO gateway as the Trunk, you can just use the default Trunk 888 and 889 as the FXO’s register number.

• For the FXO gateway, you may just only configure Trunk Number, Password, Host, DialPlan, Keypad, NAT Traversal and RTP Mode.

• If you set the ITSP as the Trunk, you may need to set the following configure: Port, External Server Address and Outbound Caller ID.

• For more information, please refer to the CH5. Appendix-Application between CPE device and EZLoop® IP-PBX Enterprise SIP Server.

 Hot-Key Tran: Enable this feature will permits the calling party or called party to

transfer a call by pressing the *0 (For Blind Transfer) or *9 (For consultant

Transfer) key if the call is Between Extension and Trunk. Default is disabled.

Note:

 If you enable this feature in Trunk page, we suggest you also enable Hot-Key

Tran of IP PBX page.

 Please note that if this option is used, the RTP Mode will always be Routed

Mode, as EZLoop® IP-PBX Enterprise SIP Server needs to monitor the call to detect when the caller presses the *0 or *9 key.

 Music RBT: Provides music to the calling party until the call is answered

Press Apply to save configuration, or press Cancel to quit configuration.

Note:

• For more information about Trunk setting, please refer to CH3. Full WEB Configurations.

CH3. Full Web Configurations

After Login EZLoop® IP-PBX Enterprise SIP Server will see screen as below, and

there are four main categories, user can click on each category to extend detail items.

 Configuration: Include all telephony configuration of EZLoop® IP-PBX Enterprise SIP

Server.

• IP PBX

• Extension

• Trunk

• SIP Trunk Reg.

• Routing Table

• Dial Group

• Speed Dial

• Others

 Information: To show related information.

• Subscriber

 Management: Include all system management of EZLoop® IP-PBX Enterprise SIP

Server.

• Network

• TimeZone

• SMTP Setting

• User Account

• Firmware Upload

• Music Upload

• Import Setting

• Export Setting

• Flash Clean

 Reboot System: To reboot system of EZLoop® IP-PBX Enterprise SIP Server.

3.1 Configuration

User can set EZLoop® IP-PBX Enterprise SIP Server telephony configuration under

Configuration category.

3.1.1 IP PBX

Enter Configuration  IP PBX to configure PBX data.

SIP Setting

 IP-PBX Realm: Configure Realm of EZLoop® IP-PBX Enterprise SIP Server. This

parameter is essential when there is more than one EZLoop® IP-PBX Enterprise SIP Server, and user wants to have inter-calls between EZLoop® IP-PBX Enterprise SIP Servers. Please refer to SIP Trunk configuration.

 Proxy Port: These optional parameters allow you to control the port on which you

wish the EZLoop® IP-PBX Enterprise SIP Server to accept SIP connections. Default is 5060.

 RTP Port Start: The voice media will use RTP as the transport protocol. You can

define the RTP port range that EZLoop® IP-PBX Enterprise SIP Server opened. Default start port is 10000.

 RTP Port End: The voice media will use RTP as the transport protocol. You can

define the RTP port range that EZLoop® IP-PBX Enterprise SIP Server opened. Default end port is 20000.

Note:

• Default RTP port range is 10000 to 20000 and default proxy port is 5060. If your EZLoop® IP-PBX Enterprise SIP Server is behind a firewall, please make sure you have already open the RTP port (10000-20000) and proxy port (5060). And you should also make sure the proxy port (5060) has already mapped to EZLoop® IP-PBX Enterprise SIP Server.

PBX Setting

 Operator: Assign operator access code. When caller dial in EZLoop® IP-PBX

Enterprise SIP Server, press this assigned code can reach operator.

 to EXT: Assign operator ‘s extension number. When caller press operator’s access

code, EZLoop® IP-PBX Enterprise SIP Server will transfer this call to the assigned Extension.

 CDR Mode: Chose the mode for CDR. You can disable the CDR or send the CDR

record to a certain CDR server. You can also store the CDR records within EZLoop® IP-PBX Enterprise SIP Server.

• Disable: Chose this one to Disable CDR function.

• RealTime: You can install a CDR program to collect and store CDR records. The

CDR program is proprietary. For more information about such CDR program, please contact with your contact window of .

Note:

 If you already setup the CDR program in a PC. Please also input the PC’s IP

address to [CDR-Server IP] field. Every 5 seconds, EZLoop® IP-PBX Enterprise SIP Server will send a CDR record to CDR-Server. And CDRServer will collect such records as a CSV file.

• Storage: If you do not prepare a PC as a CDR server. You can also define the

CDR Mode to Storage. EZLoop® IP-PBX Enterprise SIP Server will store the CDR records within itself.

Note:

 If you chose the CDR Mode to Store, you can download the CDR file by

pressing Export button of Export CDR field. When you export CDR files, EZLoop® IP-PBX Enterprise SIP Server will clean the CDR record from it. EZLoop® IP-PBX Enterprise SIP Server can only store 500 CDR records within itself. If you do not export the CDR file but the records is over than 500, the oldest one will be instead by newest CDR record.

 CDR-Server IP: If you choose the CDR Mode to RealTime, here you can input the IP

address of CDR server which you installed the CDR program.

 Export CDR: If you chose the CDR Mode to Storage, you can press Export button to

download the CDR file. The CDR file is within a CSV format.

 Ext Ring Time: This field defines the timeout value if the call is between Extension

and Extension. Default is 20 seconds.

 Out Ring Time: This field defines the timeout value if the call is from Extension to

outside (define by routing table). Default is no limitation.

 Hot-Key Tran: User can enable or disable Hot-key transfer function. If the call is

establish between Extensions. Enable this feature will permits the calling party or called party to transfer a call by pressing the *0 (For Blind Transfer) or *9 (For

consultant Transfer) key. Default is disabled.

Note:

• Please note that if this option is used, the RTP Mode will always be Routed Mode, as EZLoop® IP-PBX Enterprise SIP Server needs to monitor the call to detect when the caller presses the *0 or *9 key.

 Music RBT: Enabling this option will provide music to the calling party until the call is

answered.

System Announcement

 Scheduled Greeting: You can define a business time for the company

announcement.

 Office Hour AM: Define the work time for AM. Default is 09:00-11:59. The sound

file for greeting of work time is: greeting-day.gsm. You can change it by dialing to **111.

 Lunch Break: Define the time for lunch break. Default is 12:00-12:59. The

sound file for greeting of lunch break is: greeting-noon.gsm. You can change it by dialing to **112.

 Office Hour PM: Define the work time for PM. Default is 13:00-17:59. The sound

file for greeting of work time is: greeting-day.gsm. You can change it by dialing to **111.

 Day: Define the work days. Default is Monday to Friday.

Note:

• If the time is a non-working time, the greeting files will be greeting-night.gsm. You can change it by dialing to **113.

• For more information about the greeting record, please refer to user manual:

4.1.3 How to record the other announcements

 Fixed Greeting: You can enforce the company announcement as a fixed greeting.

 Disable: If you disable the Fixed Greeting, the announcement will base on the

above work time setting.

 Working Time: Chose this one will enforce the company announcement as a

greeting for work time: greeting-day.gsm.

 Non-working Time: Chose this one will enforce the company announcement as

a greeting for non-work time: greeting-night.gsm.

 Special Greeting: Chose this one will enforce the company announcement as a

special greeting: greeting-temporary.gsm. You can record it by dialing **115.

Behind NAT

 Behind NAT: If your EZLoop® IP-PBX Enterprise SIP Server is behind NAT, we

strongly suggest u to enable Behind NAT to avoid some unexpected issue, such as “one way voice”.

 External IP: If you input External IP, EZLoop® IP-PBX Enterprise SIP Server will take

that IP address as its argument. If EZLoop® IP-PBX Enterprise SIP Server is behind NAT, the SIP header will normally use the private IP address assigned to the server. The remote device will not know how to route back to this address; thus, it must be replaced with a valid, routable address.

 External Host: External Host takes a fully qualified domain name as its argument. If

EZLoop® IP-PBX Enterprise SIP Server is behind NAT, the SIP header will normally use the private IP address assigned to the server. If you set this option, EZLoop® IPPBX Enterprise SIP Server will perform periodic DNS lookups on the hostname and replace the private IP address with the IP address returned from the DNS lookup.

Note:

• You should not set both of External IP and External Host together; otherwise there will be some unexpected problems appeared. That means you can only chose one for External IP or External Host for “Behind NAT”

 Local Net: Local Net is used to tell EZLoop® IP-PBX Enterprise SIP Server which IP

addresses are considered local. If one of caller or callee is not under Local Net, EZLoop® IP-PBX Enterprise SIP Server will set the address in the SIP header that can be translated to that specified by External IP or the IP address can be looked up with External Host. The format will be IP/ Subnet Mask. Example: 192.168.1.0/

255.255.255.0

3.1.2 Extension

User has to set Extension account for other device to register on EZLoop® IP-PBX

Enterprise SIP Server.