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08

E series DECT Cordless VoIP Phone | User Manual

E -

series DECT VoIP Phone Users Manual

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E series DECT Cordless VoIP Phone | User Manual

Table of Contents

Function ---------------------------------------------------------------------------------------------------------

2  

Standard and Protocols ------------------------------------------------------------------------------------

2  

1.1

Overview of Hardware ----------------------------------------------------------------------------------------------------------

 

1.1.1

---------------------------------------------------------------------------------------------------------------------------------------------------- 2

1.1.2

---------------------------------------------------------------------------------------------------------------------------------------------------- 2

1.2

Overview of Software -----------------------------------------------------------------------------------------------------------

 

 

 

2

. Base Station keys for E series -----------------------------------------------------------------------

4  

2.1

Function table of keystrokes ------------------------------------------------------------------------------------------------- 4

 

2.1.1

E Series: Single Line SIP ---------------------------------------------------------------------------------------------------------------- 4

2.1.2

Call volume control ------------------------------------------------------------------------------------------------------------------------ 4

2.1.3

Hold Function ------------------------------------------------------------------------------------------------------------------------------- 4

2.1.4

Redial Function ----------------------------------------------------------------------------------------------------------------------------- 4

2.1.5 MWI – Message Waiting Indication ---------------------------------------------------------------------------------------------------- 4

2.1.6

Page – Remote DECT Handsets ------------------------------------------------------------------------------------------------------ 4

2.1.7

Speed Dial Keys ----------------------------------------------------------------------------------------------------------------------------- 4

 

 

 

 

 

 

 

3

. Web browser User Interface to Configure Phone ----------------------------------------------

4  

3.1 Login ： ----------------------------------------------------------------------------------------------------------------------------- 4  

3.2

Current Status -------------------------------------------------------------------------------------------------------------------- 5

3.3

Network ----------------------------------------------------------------------------------------------------------------------------- 5

3.3.1

Wan Config ---------------------------------------------------------------------------------------------------------------------------------- 5

3.4

VoIP ---------------------------------------------------------------------------------------------------------------------------------- 6

3.4.1 SIP Config ------------------------------------------------------------------------------------------------------------------------------------ 6

3.5

Advance ---------------------------------------------------------------------------------------------------------------------------- 7

3.5.1

DHCP Server -------------------------------------------------------------------------------------------------------------------------------- 7

3.5.2

NAT --------------------------------------------------------------------------------------------------------------------------------------------- 7

3.5.3

STUN ------------------------------------------------------------------------------------------------------------------------------------------ 8

3.5.4

Net Service ---------------------------------------------------------------------------------------------------------------------------------- 8

3.5.5

Firewall settings ---------------------------------------------------------------------------------------------------------------------------- 8

 

 

 

 

 

 

 

 

 

 

 

 

3.5.6

VLAN Configuration ----------------------------------------------------------------------------------------------------------------------- 9

3.5.7

Digital Map ----------------------------------------------------------------------------------------------------------------------------------- 9

3.5.9

Memory Key Programming ------------------------------------------------------------------------------------------------------------- 10

3.5.9.1 Mem ory Key Special Programming – Transfer, Transfer Silent and 1 second Pause --------------------------------- 10

3.5.9.2

Call Park and Retrieve ----------------------------------------------------------------------------------------------------------------- 11

3.5.10

MMI Filter ---------------------------------------------------------------------------------------------------------------------------------- 11

3.5.11

Audio Settings ----------------------------------------------------------------------------------------------------------------------------- 11

3.5.12

VPN ----------------------------------------------------------------------------------------------------------------------------------------- 11

 

 

 

 

 

 

 

3.6

Dial Peer dial rule setting -------------------------------------------------------------------------------------------------- 12

 

3.7

Config Manage ----------------------------------------------------------------------------------------------------------------- 12

 

3.8

Update Firmware -------------------------------------------------------------------------------------------------------------- 12

 

3.8.1

Update --------------------------------------------------------------------------------------------------------------------------------------- 12

 

3.8.2

 

Auto Update ------------------------------------------------------------------------------------------------------------------------------ 13

3.9

System Manage ---------------------------------------------------------------------------------------------------------------- 13

3.9.1

Account Manage --------------------------------------------------------------------------------------------------------------------------- 13

3.9.2

Syslog config ------------------------------------------------------------------------------------------------------------------------------- 13

3.9.3 Phone Book --------------------------------------------------------------------------------------------------------------------------------- 14

3.9.5

MMI SET ------------------------------------------------------------------------------------------------------------------------------------- 14

3.9.6

Logout & Reboot -------------------------------------------------------------------------------------------------------------------------- 14

 

 

 

 

 

 

 

4

. Operating Method for Dialing -----------------------------------------------------------------------

14  

4.1

How to dial IP Phone ------------------------------------------------------------------------------------------------------- 14

 

4.2

Set the phone being connected to server ---------------------------------------------------------------------------- 14

 

4.2.1

Set the WAN interface -------------------------------------------------------------------------------------------------------------------- 14

 

4.2.2

SIP setting: ----------------------------------------------------------------------------------------------------------------------------------- 15

 

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4.2.3

IAX setting ----------------------------------------------------------------------------------------------------------------------------------- 15

4 .

3 How to use the dial rule?

--------------------------------------------------------------------------------------------------- 16

 

 

Reference – Quick Keys ----------------------------------------------------------------------------------

17  

Reference – SIP Quick Start to Register a Single Phone --------------------------------------

18  

Program your connection under the VoIP > SIP Con figuration Screen ------------------------------------------------------------- 18

Speed dial and Touchlite Message keys are programmed under Advance > Memory Key ----------------------------------- 19

Reset to Factory Default ------------------------------------------------------------------------------------------------------------------------- 19

 

 

 

Reference – DECT SIP Handset Mana gement ----------------------------------------------------

20  

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E series DECT Cordless VoIP Phone | User Manual

Function

• Provide a Backup SIP Server

•

Support NAT (Network Address Translation) , Firewall

•

Support DHCP for phone to accept IP via WAN port or assign IP address via LAN port

•

Support PPPoE （ used while connecting ADSL ， cable modem ）

•

U pdate the firmware and/or configuration file program through HTTP ,FTP and TFTP

• Check the dynamic voice; Soft the noise; Buffer technique of voice

•

Hold Function

• Hotline Function (dial a specific number as soon as going off hook, or set a delay before dialing)

•

Speed dial

• Call forward, Three way conference call

•

Caller ID display

•

DND (Do Not Disturb), Black List, Limit List

•

Auto answer.

•

Set through standard Web Browser

• Remote Management Function

•

Classificat ion management for common user’s password and superuser’s password.

Standard and Protocols

•

IEEE 802.3

/ 802.3

u 10 Base T / 100 Base TX

•

PPPoE

• DHCP Client and Server

•

Support G.

711 a/u,G 729 , G 7231 5.3/6.3

audio Codec

•

SIP RFC 3261 , RFC 2543

•

Support IAX 2

• TCP/I P: Internet transfer and control protocol (Transport Control Protocol/Internet Protocol)

•

RTP: Real time Transport Protocol

•

RTCP: Real time Control Protocol

•

VAD/CNG save bandwidth (Voice Activity Detection/

•

Telnet: Internet's remote login protocol

• DNS: Doma in Name Server

•

TFTP: Trivial File Transfer Protocol

•

Introduction

•

This is t he user manual for the E series . Some configuration is necessary before using the E series phone .

This manual will illustrate how to set up the phone through keyboard and web User Interface (UI) .

1.

1

Overview of Hardware

1.1.1

The E Series cordless DECT SIP models E 103 (single line) and E 203 (two line) have two RJ 45 network interface s support ing 10/100M Ethernet. The default WAN interface is a DHCP Client server. User s connect the WAN interface to a

POE 802.3af Layer 2 switch to power the phone.

LAN port has a web bridge mode, and can bridge the LAN and WAN into the same network.

You may use the administrator’s user name “admin” and password “admin” to log in and set.

1.1.2

Only the WAN interface support s P ower O ver E thernet .

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1.2

Overview of Software

Network Protocol

•

SIP v 1 (RFC 2543

• V 2 (RFC 3261 )

)

•

IP/TCP/UDP/RTP/RTCP

•

IP/ICMP/ARP/RARP/SNTP

Codec

• G.711

： 64 K bit/s(PCM)

•

G.723.1

： 63 k/ 5.3

k bit/s

•

G.726

： 16 k/ 24 k/ 32 k/ 40 k bit/s(ADPCM)

Voice Quality

• VAD ： Voice activity detection

•

CNG ： Comfortable noise generator

• LEC ： Line echo canceller

Call Function

• Call Hold

•

•

Call Waiting

• Call Forward

3 way conference

Tone

•

Ring Tone

• Ring Back Tone

Phone Function

•

Volume Adjustment

• Speed dial key

• Phonebook

IP Assignment

•

IP (Static IP)

• DHCP

•

PP PoE

Security

QoS

• HTTP 1.1 basic/digest authentication Web setup

• MD 5 for SIP authentication (RFC 2069 /RFC 2617 )

•

QoS field

NAT Traversal

• STUN

Configuration

•

Web Browser

• Console/Telnet

• Keypad

DTMF

•

DTMF RELAY

• DTMF RFC 2833

•

DTMF SIP Info

SIP Server

• Sup port two SIP server working at the same time

• Provide a Backup SIP Server

Firmware Upgrade

• TFTP • HTTP

•

TFTP Client/DHCP Client/PPPOE Client

•

Telnet/HTTP Server

• DNS Clien ts

• G.729A

： 8 k bit/s(CS ACELP)

•

G.729B

： adds VAD & CNG to G.

729

• Packet Loss Compensation

•

Adaptive Jitter Buffer

•

Dial Tone

• Busy Tone

• FTP

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2.

Base Station

key s f or E

series

2.1

Function table of keystroke

s

Name

Hold

Status

Call

Function/Display

Hold on

Line 1

Mute

Conf

Page

8

9

0

*

4

5

6

7

Redial

1

2

3

Spkr

Volume +

Volume -

M1 ~ M7

Default

Press

Call

Single Lin e VoIP

Mute voicepath

Three way conference

Press Locate DECT Handset

Call/Dialing Call the last number dialed

Call

Call

On hook

Dialing

Dialing

Dialing

Dialing

Dialing

Dialing

Dialing

Dialing

Dialing

Dialing

Dialing

Speaker

5 or 10

“1”

“2”

“3”

“4”

“5”

“6”

“7”

“8”

“9”

“0”

“*”

Increase the volume

Decrease the volume speed dial keys

# Dialing

2.1.1 E Series : Single Line SIP

It can be regarded as the first number being dialed out or the end mark for ending number

Your E series phone is a single line , SIP protocol Voice over IP phone .

Model number E 103 denotes a single line and E 203 is a two line phoneset. Both models have 7 programmable Speed dials a vailable to program via the web user interface (UI).

2.1.2 Call volume control

Press “Volume +” to increase the volume, and “Volume

“ to decrease the volume.

2.1.

3 Hold Function

Hold the current call line.

2.1.

4 Redial Fu n ction

Redial the number dialed last time.

2.1.5

MWI – Message Waiting Indication

The OneTouch message key serves as an indicator that messages are present as well as a speed dial key to dial in and retrieve those messages. If messages are present, press the flashing red MWI k ey labeled

Messages to go off hook and dial the programmed voicemail number. First, program in web UI.

2.1.6 Page – Remote DECT Handsets

Your E Series phone comes with the DECT cordless handset automatically synchronized to the base. The

Page Key is us ed to locate any cordless DECT handset that is synchronized to the base station. You may synchronize other auxiliary handsets by placing them in the base cradle for a minute before moving them to their charging cradle.

2.1.7 Speed Dial Keys

The E series SIP set has 7 speed dial keys, programmable through the web UI under Advanced >

Memory Key .

3 .

Web browser User Interface to

Configure Phone

Insert one end of net wire RJ45 to interface of network card of computer, then insert the other end to LAN int erface of the phone, and set the computer IP on the same net with the phone IP or make it gets IP automatically. Then open a web browser , input the phone IP address in the address field, you will enter web setup page of the phone.

3.1

Login

：

T he default user name and password are admin/admin and guest/guest.

3.2

Current Status

This page layout shows the work state of VoIP phone.

The network part shows the connection state of W AN interface and LAN interface and the network setting; the work state of Public SIP service of VoIP part, and here you can see the registration and whether registered to the server or not. The Phone Number part shows the telephone numbers in Private SIP server and Public

SIP server.

E series DECT Cordless VoIP Phone | User Manual

Configure to dynamic obtain IP

Enable DHCP;

If there is DHCP server in your local network, E series will automatically obtain WAN port network information from your DHCP server.

Configure PPPoE:

3.3

Network

3.3.1 Wan Config

WAN port network setting page.

Support static IP, dynamic obtain IP and PPPoE.

Configure Static IP

Enable PPPoE

PPPoE server: Enter “ ANY ” if no specified from your ITSP.

Enter PPPoE username and pin in the username and password.

E series will automatically obtain WAN port network information from your ITSP if PPPoE setting and the setup are correct.

Notice: If user accesses the IP phone through WAN port. He/She should use the new

IP address to access the IP phone when the WAN port address was changed .

3.3.2

LAN Config

 

LAN IP Netmask: Set the IP Subnet mask for the LAN

DHCP Server: Enable DHCP service in LAN port; after user changed LAN IP, phone will automatically modify DHCP Lease Table and save the configure according to IP and netmask, DHCP server configure won ’ t take effect unless you reboot the device.

NAT: Enable NAT.

Bridge Mode: Enable this option to switch to bridge mode. IP phone won ’ t assign IP for its LAN port in bridge mode and its LAN and WAN port will be in the same network. (This setting won ’ t take effect unless you save the config and reboot the device)

Enable Static;

Set the phone’s IP address in the IP Address;

Set netmask in the Netmask field;

Set router IP address in the Gateway;

DNS Domain:

Set local DNS server in the Preferred DNS and the Alternate DNS 。

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Advanced SIP Setting 3.4

VoIP

3.4.1 SIP Config

Setting page of public S IP server :

Register Server Addr:

Register address of public SIP server

Register Server Port:

Register port of public SIP server ， default port is 5060

Register Username: the phone number)

Username of your SIP account (Always the same as

Register Password: Password of your S IP account.

Proxy Server Addr: IP address of proxy SIP server (SIP provider always use the same IP for register server and proxy server, in this case you don’t need to configure the proxy server information.)

Proxy Server Port: Signal port of SIP proxy

Proxy Username: proxy server username

Proxy Password: proxy server password

Domain Realm: SIP domain, enter the sip domain if any, otherwise E ser ies will use the proxy server address as sip domain.

Local SIP port: Local SIP register port, default 5060

Phone Number: Phone number of your SIP account

Enable Register: Enable/Disable SIP register.

E series won’t send register info to SIP server if register is disabled .

Enable Message Waiting: The configuration allows/ forbids Message Waiting.

6

Register Expire Time: register expire time, default is 60 seconds. E series will auto configure this expire time to the server recommended setting if it is different from the

SIP server.

Call Forward: Please refer to Value_add_service for detail.

N o answer: If no answer, it will forward to appointed phone.

Always: The caller always forward to the appointed phone.

Forward Photo Number: call the forwarded phone number.

Detect Interval Time: Co-work with the Auto Detect Server, if Auto Detect Server is enable, E series will periodically detect if the SIP server is a vailable according this setting.

User Agent:

Encrypt Key: The particular service system decrypts of the key, matching with the server Type usage, the key provide by the particular service system supplier, default is empty

Server Type: T he particular ser vice system supplier carries out the sign and speeches to encrypt, default is common

DTMF Mode: DTM signal sending mode: support

RFC2833, DTMF_RELAY (inband audio) and SIP info

RFC Protocol Edition: Current E series SIP version. Set to

RFC 2543 if the gate need to communicate to devices

(such as CISCO5300) using the SIP 1.0. Default is RFC

3261.

3.4.2 I ax2 Config

Setting page of public IAX server :

IAX Server Addr: Register address of public IAX server

IAX Server Port: Register port of public IAX server ， default port is 4569

Account Name: Username of your SIP account (Always the same as the phone number)

Account Password: Password of your IAX account.

Local port: Signal port of local, default port is 4569 www.Cetis

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Phone Number: Phone number of your IAX account

Voice mail number: If the IAX support voice mail, but your username of the voice mail is letter s which you can not input with the ATA, then you use the number to stand for your username

Voice mail text: if IAX support voice mail, config the domain name of your mail box here.

Echo test number: If the platform support echo test, and the number is test form, the config the test number to replace the text format The echo test is to test the woring status of terminals and platform

Echo test text: echo test number in text format

Refresh time: IAX refresh time

Enable Register: enable or disable register

Enable G.729: Using G.

729 speech coding mandatory consultations

IAX2 (Default Protocol): Set IAX 2 as the default protocol, if not the system will choose SIP as default

E series DECT Cordless VoIP Phone | User Manual

Lease Time: DHCP server lease time.

Netmask: Netmask of lease table.

Gateway: Default gateway of lease table

DNS: default DNS server of lease table.

DNS Relay: enable DNS relay function.

U ser may use below setting to add a new lease table.

N otice: This setting won ’ t take effect unless you save the config and reboot the device

3.

5

Advance

3.5.1 DHCP Server

DHCP server manage page.

User may trace and modify DHCP server information in this page.

DHCP Lease Table ： display the IP － MAC corresponding table that the server distributed.

Lease Table Name: Lease table name.

Start IP: Start IP of lease table.

End IP: End IP of lease table. N etwork device connecting to the E series LAN port can dynamic obtain the IP in the range between start IP and end IP.

3.5.2 NAT

Advance NAT setting. M aximum 10 items for TCP and

UDP port mapping.

DHCP Lease Table ： Show IP — MAC corresponding table assigned by DHCP server.

IPSec ALG:

FTP ALG:

Enable/Disable IPSec ALG;

Enable/Disable FTP ALG;

PPTP ALG: Enable/Disable PPTP ALG;

Transfer Type: Transfer type using port mapping.

Inside IP: LAN device IP for port mapping.

Inside Port: LAN device port for port mapping.

Outside Port: WAN port for port mapping.

C lick Add to add new port mapping item and Delete to delete current port mapping item.

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DMZ Config ：

3.5.3 STUN

This page is used to set the private sip server, stun server, and back up sip server information.

STUN Server setting: SIP STUN is used to realize SIP penetrates through NAT, when the phone configures IP and port of STUN server (default is 3478) and select

Enable SIP Stun, common SIP server can be used to realize the phone to penetrate through NAT. In this way, If you have common SIP proxy and STUN server parked public network, it is all right, but STUN only support three NAT ways: FULL CONE, restricted, port restricted;

STUN Server Addr: configure stun server address;

STUN Server Port: configure stun server port default

3478

STUN Effect Time: stun detect NAT type circle, unit: minute.

Local SIP Port: The SIP port of this phone

Load ： Load the choices of SIP line.

Use Stun: Stun. Set the Stun that allows/forbids use user setting.

3.5.4 Net Service

HTTP Port: configure HTTP transfer port; default is 80.

User may change this port to enhance system ’ s security. W hen this port is changed, please use http://xxx.xxx.xxx.xxx:xxxx/ to reconnect.

Telnet Port: configure telnet transfer port, default is 23.

RTP Initial Port : RTP initial port.

RTP Port Quantity: Maximum RTP port quantity, default is 200

Notice:

Settings in this page won’t take effect unless save and reboot the device.

If you need to change telnet port or HTTP port, please use the port greater than 1024, because ports under

1024 is system remain ports.

HTTP service if HTTP is set to 0.

8

3.5.5 Firewall settings

Firewall setting page ： User may set up firewall to prevent un authorized Internet users from accessing private networks connected to the Internet (input rule), or prevent unauthorized private network devices to access the internet.

Access list support two type limits: input_access limit or output_access limit. Each type supports 10 items maximum.

E series firewall filter is base d upon the WAN port. So the source address or input destination address should be WAN port IP address. www.Cetis

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Configuration:

I n _access enable enable in_access rule

Out_access enable enable out_access rule

Input/Output : current adding is input rule or output rule.

Deny/Permit: current adding is deny rule or permit rule.

Protocol Type: protocol using in this rule: TCP/IP/

ICMP/UDP.

Port Range : port range if this rule

Src Addr : source address. It is can be single IP address or network address.

Dest Addr: destination address. It is can be IP address or network address.

Src Mask: source address mask. Indicate the source is dedicate IP if set to 255.255.255.255. Otherwise is network ID

Des Mask: Destination address mask. Indicate the source is dedicate IP if set to 255.255.255.255.

Otherwise is network ID

DiffServ Enable : If enable the VLAN service, it ind icates use DSCP mode to realize three layers QoS. This moment, the DSCP of SIP signals which between E series Phone and MGC. It will use Class Selector 5

(The value is 0xA0). And the DSCP of mediums information (In RTP packets) would be used th e values of DiffServ Value field .

DiffServ Value: The value range ：

0x28,0x30,0x38,0x48,0x50,0x58,0x68,0x70,0x78,0x88,

0x90,0x98,0xb8.default is 0xb8 ,o xb8 stands for best fast transmission; 28 3 8 is guarantee for the transmission priority for the 1st rank , 48 58 is guarantee for the transmission priority for the 2nd rank,

68 78 is guarantee for the transmission priority for the

3rd rank, 88 98 is guarantee for the transmission priority for the 4th rank .

802 ． IP Priority: The priority of 802.ip

3.5.6 VLAN Configuration

E series phone s implement QoS (Quality of Service) based on 802.1p, The QoS is used to mark the network communication priority in the data link/MAC sub-layer.

E series will sort the packets using the QoS and sends it to the destination.

VLAN Enable: E nable the VLAN service to separate voice and data transmission on distinct VLANs .

VLAN ID: Dispose VLAN ID is adding a Tag header after realize enable the VLAN function. The realized voice packets transfer at the same VLAN. The prerequisite is it must the same as VLAN of upper switch. The value range are 1 ~ 4094 .

9

3.5.7 Digital Map

Digit map is a set of rules to determine when the user has finished dialing.

E series support s below digital map:

Digital Map is based on some rules to judge when user end their dialing and send the number to the server. E series support th e following digital map:

End With “ # ” : Use # as the end of dialing.

Fixed Length : When the length of the dialing match, the call will be sent.

Timeout: Specify the timeout of the last dial digit.

T he call will be sent after timeout

Prefix: User define dig ital map

-

[ ] represents the range of digit, can be a range such as [1 4], or use comma such as [1,3,5], or use a list such as [234] x

Tn represents any one digit between 0~9 represents the last digit timeout. n represents the time from 0~9 second, it is necessary. Tn must be the last two digits in the entry. If Tn is not included in the entry, we use T0 as default, it means system will send the number immediately if the number matches the entry.

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8[2-8]xxxxx All number from 8200000 to 8899999 will be sent immediately.

955xx5 digits numbers begin with 9 will be sent immediately .

10060 Number 10060 will be sent will be immediately

22xxxxx T1 7 digits numbers begin with 22 will be sent after one second

39[3,9]xxxx, 7 digits numbers begin with 393 or 399 will be sent immediately .

3.5.9 Memory Key Programming

This page layout shows the number setting of Speeddial key s .

There are several special fu nctions that can be programmed into Memory Keys, such as Transfer,

Transfer Silent, and Call Park/Retrieve .

3.5.8 Call Service Settings

User s configure the value add service such as hotline, call forward, call transfer, call waiting, 3-way conference call, auto-answer, etc in this page .

Hotline: configure hotline number . E series immediately dials this number after hook-off if it is set.

Auto Answer: Enable/disable auto answer function.

No Disturb: DND, do not disturb, enable this option to refuse any calls.

Ban Outgoing: Enable this to ban outgoing calls.

Enable Call Transfer: Please refer to Value_add_service for detail.

Enable Call Waiting: Enable/disable Call Waiting

Enable Three Way Call: Please refer to

Value_add_service for detail.

Accept Any Call: If this option is disabled, E series refuse the incoming call when the called number is different from E series ’ s phone number.

No Answer Time: no answer call forward time setting.

Black List: incoming call in these phone numbers will be refused.

Limit List: outgoing calls with these phone numbers will be refused

3.5.9.1

pro

Memory Key Special Programming

Transfer, Transfer Silent and 1 -

– second Pause

Transfer occurs when there is an active/live call in gress. If no call is active, the ‘T’ is ignored, and the memory key will just call the programmed number

(1107). In the settings below, Memory 1 (T1107) will transfer to extension 1107 when pressed. The user will hear the DTMF tones as they are transferred to 1107 .

In the example of Memory 2 , (TS 1107 ) will transfer

“silently” to extension 1107 . ‘Silent’ means the user will not hear the DTMF tones.

In Memory 3 , the transfer to 1107 will occur, after the connection is established, a pause will happen t hen digits are sent. P= 1 second pause, so PPP = 3 seconds, then the digits ( 12345 ) are sent.

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3.5.9.2

Call Park and Retrieve

Call Park/Retrieve allows the phone’s user to send a call to a ‘Park’ location (setup within the PBX), where another user may then ‘Retrieve’ the call from a different phone.

It is programmed as follows in two steps:

1 . Under SIP Config => Advanced Set you will find Park

Mode. Default is Off, and Park 1 is enabled or On.

3.5.1

1 Audio Settings

CODEC: select the prefer CODEC; support ulaw, alaw ,

G729 and G7231 5.3/6.3

Signal Standard: Signal standard for different area.

Handdown Time: hand down detect time.

Input Volume: Handset in volume.

Output Volume: Handset out volume.

Handfree Volume: Hand free volume

G729 Payload Length: G729 payload length

VAD: Enable/disable Voice Activity Detection

2. Under Advanced => Memory Key, program the

Memory Ke y HdActive to Transfer Silent the call to the

Park Location ( 70 in the example below). Eg:TS 70

Also program Memory Key HdIdle to Retrieve from the

Parked Location. In this case TS71. The Park and

Retrieve numbers (70 and 71) are arbitrary, and subject t o the programmer of the PBX. They are used her as an example only.

3.5.10 MMI Filter

MMI filter is used to make access limit to

E series phone.

W hen MMI filter is enable. Only IP address within the start IP and end IP can access E series phone.

11

3.5.12 VPN

This page is VPN setting page, the IP phone support the VPN with UDP and L2TP protocol .The parameters is as below

VPN IP : A fter VPN registered successfully, VPN server will give an IP aggress to the terminal . If there is a IP address shown on terminal ( except for 0.0.0.0) , it means your VPN has registered UDP Tunnel

VPN Server Addr : register to the address of VPN server

VPN Server Port: r egister to the port of VPN server

Server Group ID: the group ID of UDP VPN

Server Area Code : the area code VPN server L2TP

VPN Server Addr: register to the address of VPN server www.Cetis

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VPN User Name : L2TP VPN username

VPN Password : L2TP VPN password

UDPTunnel : use the UDP to visit VPN

L2TP : use the L2TP to visit VPN

Enable VPN : Enable the VPN server , you must choose

UDP or L2TP type in advance

Notice: At the present, L2TP only support L2TP VPN server under Linux, UDP only support a private UDP

VPN server.

E series DECT Cordless VoIP Phone | User Manual window, then type the config file name in the File name

( the file type is text file)

Update Configuration: Update the current configuration through configuration files.

Notice: clear config in admin mode, all settings restores to factory default; clear config in guest modem, all settings except sip, advance sip restore to factory default.

3.6

Dial-Peer dial rule setting

P lease refer to How to use dial rule for detail.

3.7 Config Manage

Save Config: save current settings.

Clear Config: restore to default settings.

B ackup Config: Backup the config file, via point the right key of mouseà save target as … .à will pop a save

12

3.8

Update Firmware

3.8.1 Update

Web Update ：

Update the application or configuration files of the phone. T he application document is .z format, and the configuration files are .cfg or .txt file format.

Through clicking on the "browse" button to open the upgrade file or configuration file , then click on "Update" button. After the upgrade, E series will automatically restart.

FTP Update:

Upload/download the configure file with FTP or TFTP server, or download firmware from FTP or TFTP server

Back up configure file to your FTP/TFTP server. configure use .cfg extension.

The Type includes two parts of config file export and config file import

C onfig file export: export the config file

Config file import: import the config file

E series phone s support FTP and TFTP auto update, the gateway will auto obtain the configure file from your update server if configured. T o obtain the original configure file, you can use the FTP/TFTP back up as describe above. C onfigure file using module structure, user may remain the concerned modules and remove other modules. Put the configure file in the root directory of update server when finish editing. www.Cetis

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E series DECT Cordless VoIP Phone | User Manual

3.8.2 Auto Update

Current Version : the system will display the current version number

Server Address: FTP/TFTP server address

Username: FTP server user name

Password: FTP server password

Config File Name: T he name of configuration file

Config Encrypt Key: T he encrypt key of confirmation file

Protocol Type: T he protocol type that used for upgrading

Update Interval Time: T he interval time that the terminals search for new configuration file.

Update Mode: auto provision mode; Disable: not auto update ， Update after reboot: auto update after reboot ， Update at time interval: auto update after a certain time

C onfigure file version was in the <<VOIP CONFIG

FILE>> Version 1.0007 and <GLOBLE CONFIG

MODULE> ConfFile Version

For instance:

Gateway original version is:

<<VOIP CONFIG FILE>>Version: 1.0000

<GLOBLE CONFIG MODULE> ConfFile Version ： 6

User may edit the configure file version to:

<<VOIP CONFIG FILE>>Version: 1.0007

<GLOBLE CONFIG MODULE> ConfFile Version ： 7

3.9 System Manage

3.9.1 Account Manage

S et web access account or keypad password of E series .

The factory default is 123

3.9.2 S yslog config

S et the system log

S erver IP: set the syslog server address

Server Port: set the syslog server port

MGR Log Level: set the MGR log level

SIP Log Level: set the SIP log level

IAX2 Log Level: set the IAX2 log level

P lease click “ apply ” after setting

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3.9.3 P hone Book

E series DECT Cordless VoIP Phone | User Manual

Reboot Phone ： logout the entry, and reboot the phone.

W hen user to modify any config of the phone, it will take effect after being rebooted, you can enter into this layout and click “ Reboot ” . A nd the phone will be rebooted automatica lly .

Note ： Reboot IP phone, some setting needs to reboot to make it works. P lease always save config before reboot, otherwise the setting will return to previous setting.

3.9.4 Time Set

This page layout is the setting of time of phone.

Server: type the IP address of time server

Timezone: select correct time zone in list box

T imeout: longest respon s e time for SNTP

Daylight: Daylight saving time

SNTP: select SNTP server

12 Hours Format: select 12 hours format

Manual Config: The time setting

3.9.5 MMI SET

S et the greeting information on LCD.

4

.

Operating Method for Dialing

4.1 How to dial IP Phone

Yon can make a call after being made a proper setting on your phone. P lease confirm whether all the net wires are connected correctly .

If you want to make a call, you can make it after dialing the number and then pressing “ # ” .

Y o u can find IP add r e ss by the menu.

Modifying the IP address of the computer, and making it the same net with the E series .

I nput ting the IP address of the phone in the browser , and then you can visit the setting s after press ing the

Enter key; super user account is admin/admin; comm on user account is guest/guest.

4.2 Set the phone being connected to server

4.2.1 Set the WAN interface

The method of connecting to the Network à WAN Config layout phone of the net port:

E series can co nnect to the Internet by using S tatic IP,

DHCP IP, or PPP oE dialing.

3.9.6 Logout & Reboot

Logout ： logout the Web entry.

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E series DECT Cordless VoIP Phone | User Manual

4 .2.2 SIP setting:

Configure Static IP:

choose static;

fill in the IP address of E series in the IP address;

fill in the subnet mask in Netmask;

fill in the router address or up Gateway address in the Gateway;

fill in the local DNS server ad dress in the Primary

DNS and Alter DNS respectively.

Use the configure to dynamic obtain IP to get IP address :

choose DHCP option.

Now, if the network has DHCP server, then E series will get IP address, Netmask, Gateway, Primary DNS and Alter DNS from this DHCP server automatically.

Use PPPoE dialing to connect the Internet:

Enter into the VoIP à SIP Config to set the layout config and sip account information:

Register Server Addr: Registe r address of public SIP server

Register Server Port: Register port of public SIP server . default port is 5060

Register Username: Username of your SIP account

(Always the same as the phone number)

Register Password: Password of your SIP account.

Phone Number: Phone number of your SIP account

choose Enable Register ；

You can dial VoIP phone when the WAN interface and

IAX protocol are being set correctly.

4.2.3 IAX settin g

choose PPPoE option.

please fill in the account and password which

PPPoE have dialed in the PPPoE Username and

Password.

So E series could connect the Internet th rough PPPoE dialing, and automatically get IP address, Netmask,

Gateway, Primary DNS and Alter DNS and so on.

15

IAX Server Addr: Register address of public IAX server

IAX Server Port: Re gister port of public IAX server , default port is 4569

Account Name: Username of your SIP account (Always www.Cetis

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the same as the phone number)

Account Password: Password of your IAX account.

Local port: Signal port of local, default port is 4569

Phone N umber: Phone number of your IAX account

choose Enable Register ；

if you use IAX account to make a call, please choose IAX （ De f ault Protocol ） , if you fail to choose it, then you can use SIP account to make a call again.

if you use G..

729 to arrange it ,please choose

Enable G..

729

You can dial VoIP phone when the WAN interface and

IAX protocol.

Note: please choose Save Config in the Config Manage after setting the information, or the existing setting information will be failed after rebooting..

4.3

How to use the dial rule?

E series provide fle xible dial rule, with different dial rule configure, user can easily implement the following function:

Replace, delete or add prefix of the dial number.

Make direct IP to IP call

Place the call to different servers according the

prefix.

You can click “Ad d” to add a new dial rule. Below is the detail setting of the dial rule:

Phone Number: The Number suit for this dial rule, can be set as full match or prefix match. Full match means that if the number user dialed is completely the same as this number, the call will use this dial rule. Prefix match means that if prefix of the number that the user dials is the same as the prefix, the call will use this dial rule, to distinguish from the full match case, you need to add

“T” after the prefix number in the phon e number setting.

Call Mode: support SIP.

Destination (optional): call destination, can be IP or domain. Default is 0.0.0.0, in this case the call will be routed to the Public SIP server. If you set the destination to 255.255.255.255, then the call will be routed to the private SIP server. Also you can key other address here to make direct IP calls

Port (optional): Configure the port of the destination, default is 5060 in SIP

Alias (optional): Set up the Alias. We support four Alias as below. Alias nee d to co work with the Del Length: add:xxx, add prefix to the phone number, can set to reduce the dial length.

E series DECT Cordless VoIP Phone | User Manual all: xxx, replace the phone number with the xxx, can use as speed dial function.

del, delete the first N numbers. N is set in the

Del Le ngth rep:xxx ， replace the first N numbers. N is set call 86633 8215555, then you can set the phone number in the dial rule as 0633T, and set the Alias as rep:86633, and set the Del

Length to 4. Then all calls begin with 0633 will be changed to 86633 xxxxxxxx.

Suf fix (optional): Configure suffix, show no suffix if not set Instance

2T rule: If the call starts with 2 , the first 2 will be deleted, and the rest number will be sent to private SIP server.

3T rule: If the call starts with 3, the first 3 will be dele ted, and the rest number with be sent to public SIP server.

123 rule: Dial 123 and will send 06332221015 to your server. Used as speed dial function.

0T rule: If the calls is begin with 0, the first 0 will be replace d by 86. Means that if you dial 063322 21015 and AG 188 will send 866332221015 to your server.

11 rule: when you dial 11 , the call with send to

192.168.0.11

, suit for LAN application without set up a sip server.

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Reference – Quick Keys

The TeleMatrix family of SIP phones has implemented an innovative way to determine various states of the SIP phone and the network it is attached to via the phone’s dialpad. By pressing the * key twice in succession followed by a series of numbers

E series DECT Cordless VoIP Phone | User Manual

(generally matched to the command’s corresponding letters.)

Many of these features are newly added in Firmware 1.8.0

and higher.

Show IP address **47# (**IP#)

Description: Keying **47# causes the phone to read out the IP a ddress currently assigned. If the phone has an

LCD panel, it should be displayed there. If the phone has no display panel, or the address cannot be displayed, it should be read out audibly on the speaker (or handset if off hook.)

Show VLAN - **85# (**VL#)

Description: When entered respond with the Voice VLAN ID. If VLAN is disabled “Zero” to indicate “off” is spoken.

Set VLAN - **87# (**87*password*VLANid#)

Description: This allows the user to set the VLAN ID. The keypad password is also required. To set the VLAN

ID when the password is 9382 and the VLAN ID is 30 , enter "** 87 * 123 * 301# ". When a VLAN ID is set, the phone should set VLAN Enabled and the Voice VLAN ID. If necessary the LAN should be set to Bridge mode also.

Show Firmware Version **39# (**FW #)

Description: When ** 39# is entered, the phone will display or audibly respond with the firmware version number such as " 1.7.240.235

".

Show TFTP Server Address **83# (**TF#)

Description: When entered, the phone responds with the IP address of the TFTP server. This should be the TFTP server address being actually used. It might come from the configuration file, option 66 , option 125 or option 43 .

Re program New ConfigID **77*password*ConfigID# (**RP*xxx*yyyy#)

Description: After entering **77*, the u ser must enter the Keypad Password followed by a * then the ConfigID of the new config file, followed by #. If the password (in the existing config on the phone is 123, and the new config file is 1234.3300ip.txt), then the user would enter **77*123*1234#. The user should NOT be prompted for the password. The phone will then download the new config file 1234.3300

ip.txt.

Reset phone to Factory Default – **33*<password># = (**DE*123#)

Description: After entering **33*123#, the phone will reset to factory defa ult, which implies that there is no programming information current, and the phone will prompt for a ConfigID at reboot.

Re boot phone (without reset) – **72# (**RB#)

Description: After entering 72# , the phone will reboot

SIP TeleWorker Config File Downl oad – **89*<tftp server IP address>*<ConfigID>#

Description: To download a config file for a phone remotely, this **TW code is used for designating a tftp server

IP address and configID to download.

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Reference –

SIP

Quick Start to Register a Single Phone

The

Cetis, Inc SIP sets are SIP only endpoints powered only by POE/802.3af and will acquire an

IP address when met by a DHCP server. Begin by plugging your POE Ethernet cable into the WAN port. To retrieve an IP address, press * * 4(I) 7(P) #. The phone’s IP address will be displayed on the LCD or spoken by audio through the speaker. Pointing your web browser to that IP address.

You will be presented with a login screen.

Login = admin

This is the Default Current Status

Page

P

assword = admin

The scope of this document is to provide the basic screenshots necessary to quickly configure the guest phone for basic registration and show the user where to program voicemail retrieval keys and s peed dials.

Our example SIP phone is programmed as extension 1001. The firmware version of the phone is circled in red.

Program your connection under the VoIP > SIP Configuration Screen

Server address of the IP/PBX, extension number/account name, phone number and display name, enable register and enable message waiting indication are all programmed here .

E series DECT Cordless VoIP Phone | User Manual

Speed dial and Touchlite Message keys are programmed under Advance > Memory Key

MWI Number is the field for programming the Red Message Waiting Indicator Touchlite.

In this example, it is programmed to dial number (7001) on the IP/PBX for retrieving voicemail messages.

Manual Programming Programming the 3300 IP via the LCD and Submerged Keys

There are 2 submerged keys (above the 10 speed dial k eys/underneath the faceplate). The left (store) key is used to store speed dials as well as enter the internal menu to check status and program the phone.

1. Speed Dial : To program a speed dial, Press the store key … press the digits you want stored … press a speed dial key.

2. Internal Menu : To enter the phone’s internal menu, hold down the left (store) key for three seconds. At that time, you will be prompted to enter the password (default password = 123) then press the other key (right key = flash) to enter the LCD navigation menu.

Vol up key navigates up

Vol Down key navigates down

Right/Flash key is Enter/Select

Left/Store key is Escape (back out of that menu option)

To determine the IP address assigned to the phone, Navigate to Config > Network > WAN >

Status using the keys and process outlined above.

Other useful documents are Reference IP DECT

Handset Registration and De Registration &

Reference 1.8.0 Quick Keys

Reset to Factory Default

To reset the phone to the factory default settings, press “#” during the startup procedure (At power up you can see a black progress bar). The phone will enter into “post” mode, then input * # 1 6 8 Then you will see “clearing conf” on the screen, next you see “conf reset”. Now you have reset to the default settings. Lastly, cycle power (remove and replace WAN power cable) to restart the phone with factory default settings.

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Reference – DECT SIP Handset

Management

The VoIP DECT cordless phone has a base station that is powered via Power Over Ethernet

(POE) 802.3af. The max power consumption is

3.5W, thus falling into the POE class spectrum of

Class 2. The auxiliary handset stations for E -

Series are POE, while the 9600 series and are powered via standard electrical plugs. All newer models (20 11 and beyond) will auto sync by placing the handset in the charging cradle of the

SIP base station. Other older models require the manual procedure described below. ALL models can register handsets via the manual method described here.

Register One Hands et:

1. Plug the DECT SIP base into a Power

Over Ethernet (POE) Layer 2 Switch

2. Wait for ConfigID tone prompt

3. Enter ConfigID and # to retrieve file from tftp server OR Press # to bypass ConfigID

4. Press Speaker key on Base to confirm dial tone

5. Press Speaker key again to hang up

6. Install battery in remote handset

7. On the DECT SIP base – Press and Hold

Locate or Page Key until charging light

Blinks

8. On Handset – Press and hold the * key –

Listen for short beep and continue holding the * key until you hear the second longer

9.

“sing song” series of beeps.

On handset – Press on/off key and listen to confirm you have dial tone.

Register up to a total of 5 handsets:

It is possible to register up to 4 remote handset kits to the base. Repeat this process for three more hand sets (LEAVE BATTERIES IN ALL

HANDSETS)

Only 5 total handsets can be registered to a base.

When a sixth one tries to register to the base it fails. The next procedure deletes the handset registrations so that you can start again.

Delete Handset Registrati ons from DECT SIP Base:

Press and hold down the submerged (right side) “FLASH” key on the base unit for three seconds, the Speaker will give a single

“beep”, then

Press “0” on the base unit to delete all the handsets.

OR

Press “1” on the base unit to delete the first handset.

Press “2” on the base unit to delete the second handset.

Press “3” on the base unit to delete the third handset.

Press “4” on the base unit to delete the fourth handset.

Press “5” on the base unit to delete the fifth handset.

N ote : After pressing the number keys, Charge indicator light will flash, which means the handset is being deleted by the phone. When

Charge indicator light stops flashing, it means that the deletion is finished.

Changing Handset Ring Tone: 6 ring tones to choose from

Hold FNC key on handset for 2 seconds then release

Press any key 1 ~ 6 (on handset) to preview the handset ringtone – 1 is default tone

Press flash to exit selection state and save any of those 1~6 tones

Lower the ring volume on the side of the handset