Cetis IP100 Manuale utente

IP100 VoIP Phone | User Manual

www.CetisGroup.com

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IP100 VoIP Phone | Users Manual

www.CetisGroup.com

Table of Contents

Functions--------------------------------------------------------------------------------------------------------2!

Standards and Protocols ----------------------------------------------------------------------------------2!

1.1 Overview of Hardware----------------------------------------------------------------------------------------------------------2!

1.1.1---------------------------------------------------------------------------------------------------------------------------------------------------- 2!

1.1.2---------------------------------------------------------------------------------------------------------------------------------------------------- 2!

1.2 Overview of Software-----------------------------------------------------------------------------------------------------------3!

2. Handset keys of IP100 -----------------------------------------------------------------------------------4!

2.1 Function table of keystroke --------------------------------------------------------------------------------------------------4!

2.1.1 Store numbers: ----------------------------------------------------------------------------------------------------------------------------- 4!

2.1.2 Call volume control ------------------------------------------------------------------------------------------------------------------------ 4!

2.1.3 Hold Function ------------------------------------------------------------------------------------------------------------------------------- 4!

2.1.4 Redial Function ----------------------------------------------------------------------------------------------------------------------------- 4!

2.1.5 Three-way conference call--------------------------------------------------------------------------------------------------------------- 4!

2.1.6 Call Transfer--------------------------------------------------------------------------------------------------------------------------------- 4!

3. Web browser User Interface to Configure Phone ----------------------------------------------4!

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

3.2 Current state----------------------------------------------------------------------------------------------------------------------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 --------------------------------------------------------------------------------------------------------------------------------10!

3.5.10 MMI Filter ----------------------------------------------------------------------------------------------------------------------------------10!

3.5.1 1 Audio Settings ----------------------------------------------------------------------------------------------------------------------------11!

3.5.12 VPN -----------------------------------------------------------------------------------------------------------------------------------------11!

3.6 Dial-Peer dial rule setting -------------------------------------------------------------------------------------------------- 11!

3.7 Config Manage ----------------------------------------------------------------------------------------------------------------- 11!

3.8 Update Firmware -------------------------------------------------------------------------------------------------------------- 12!

3.8.1 Update ---------------------------------------------------------------------------------------------------------------------------------------12!

3.8.2!Auto Update ------------------------------------------------------------------------------------------------------------------------------12!

3.9 System Manage---------------------------------------------------------------------------------------------------------------- 13!

3.9.1 Account Manage --------------------------------------------------------------------------------------------------------------------------13!

3.9.2 Syslog config -------------------------------------------------------------------------------------------------------------------------------13!

3.9.3 Phone Book---------------------------------------------------------------------------------------------------------------------------------13!

3.9.5 MMI SET-------------------------------------------------------------------------------------------------------------------------------------13!

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!

4.2.3 IAX setting -----------------------------------------------------------------------------------------------------------------------------------15!

4.3 How to use the dial rule? --------------------------------------------------------------------------------------------------- 15!

Reference – Quick Keys---------------------------------------------------------------------------------- 17!

IP100 VoIP Phone | User Manual

www.CetisGroup.com

Reference – SIP Quick Start to Register a Single Phone-------------------------------------- 18!

Program your connection under the VoIP -> SIP Configuration Screen-------------------------------------------------------------18!

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

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

IP100 VoIP Phone | User Manual

www.CetisGroup.com

Functions

•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）

•Update 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

•Classification management for common user’s password and superuser’s password.

Standards and Protocols

•IEEE 802.3 /802.3 u 10 Base T / 100Base TX

•PPPoE

•DHCP Client and Server

•Support G.711a/u,G729, G7231 5.3/6.3 audio Codec

•SIP RFC3261, RFC 2543

•Support IAX2

•TCP/IP: 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: Domain Name Server

•TFTP: Trivial File Transfer Protocol

•Introduction

•This is the user manual for the IP100. Some configuration is necessary before using the IP100 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 two RJ-45 network interface support the 10/100M Ethernet. The default WAN interface is a DHCP Client server.

User connect the WAN interface to ADSL or switch, Lan is web-bridge mode, and bridged the LAN and WAN into the

same network. You can use the administrator’s user name “admin” and password “admin” to login and set.

1.1.2

Only the WAN interface supports the POE.

IP100 VoIP Phone | User Manual

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1.2 Overview of Software

Network Protocol

•SIP v1(RFC2543)

•V2(RFC3261)

•IP/TCP/UDP/RTP/RTCP

•IP/ICMP/ARP/RARP/SNTP

•TFTP Client/DHCP Client/PPPOE Client

•Telnet/HTTP Server

•DNS Clients

Codec

•G.711：64K bit/s(PCM)

•G.723.1：63k/5.3k bit/s

•G.726：16k/24k/32k/40k bit/s(ADPCM)

•G.729A：8k bit/s(CS-ACELP)

•G.729B：adds VAD & CNG to G.729

Voice Quality

•VAD：Voice activity detection

•CNG：Comfortable noise generator

•LEC：Line echo canceller

•Packet Loss Compensation

•Adaptive Jitter Buffer

Call Function

•Call Hold

•Call Waiting

•Call Forward

•3-way conference

Tone

•Ring Tone

•Ring Back Tone

•Dial Tone

•Busy Tone

Phone Function

•Volume Adjustment

•Speed dial key

•Phonebook

IP Assignment

•IP (Static IP)

•DHCP

•PPPoE

Security

•HTTP 1.1 basic/digest authentication Web setup

•MD5 for SIP authentication (RFC2069/RFC2617)

QoS

•QoS field

NAT Traversal

•STUN

Configuration

•Web Browser

•Console/Telnet

•Keypad

DTMF

•DTMF RELAY

•DTMF RFC 2833

•DTMF SIP Info

SIP Server

•Support two SIP server working at the same time

•Provide a Backup SIP Server

Firmware Upgrade

•TFTP

•HTTP

•FTP

IP100 VoIP Phone | User Manual

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2. Handset keys of IP100

2.1 Function table of keystroke

Name

Status

Function/Display

Hold

Call

Hold on

Store

On-hook

To enter storage mode

for speed dialing

FWD

Call

Call forward

Conf

Call

Three-way

communication

Redial

Call/Dialing

Call the number dialed

last time

Speaker

Volume +

Call

Increase the volume

Volume -

Call

Decrease the volume

M1 ~ M5

On-hook

5 speed dial numbers

Dialing

“1”

Dialing

“2”

Dialing

“3”

Dialing

“4”

Dialing

“5”

Dialing

“6”

Dialing

“7”

Dialing

“8”

Dialing

“9”

Dialing

“0”

Dialing

“*”

Dialing

It can be regarded as the

first number being dialed

out or the end mark for

ending number

2.1.1 Store numbers:

Press “STORE” key to enter into the setting state of

storing keys, and input the storing number, then press

the corresponding storing key to store the numbers.

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 Function

Redial the number dialed last time.

2.1.5 Three-way conference call

Suppose the user of the IP100 phone is A, and if user B

through VoIP phone to call user A, user B need to make

3-way conference call with user A and C, then user A

can press “Conf” key to hold the conversation with user

B, then call user C, during the conversation with C to

press down “conf” key, then it can realize three-way

conference, and at that time "Conference" will be

displayed on the screen.

2.1.6 Call Transfer

The “Call Transfer” in the “Call ServerEnable” should be

in chosen mode (the default one is chosen mode).

Suppose the user of IP100 is user A, and user B called

user A，during the communication, user B want to

make a call to user C, then user A should press FWD,

after that user A can call user C through dialing

numbers.

3. Web browser User Interface to

Configure Phone

Insert one end of net wire to interface of network card of

computer, then insert the other end to LAN interface of

the phone, and set the computer IP on the same net

with the phone IP or make it gets IP automatically. Then

open the IE, input the phone IP address in the address

field, you will enter web setting page of phone IP100.

3.1 Login：

The default user name and password are admin/admin

and guest/guest.

IP100 VoIP Phone | User Manual

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3.2 Current state

This page layout shows the work state of VoIP phone.

The network part shows the connection state of WAN

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.

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 Static;

-Set IP100’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。

Configure to dynamic obtain IP

Enable DHCP;

If there is DHCP server in your local network, IP100 will

automatically obtain WAN port network information from

your DHCP server.

Configure PPPoE:

-Enable PPPoE

-PPPoE server: Enter “ANY” if no specified from

your ITSP.

-Enter PPPoE username and pin in the username

and password.

IP100 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 Netmask 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)

IP100 VoIP Phone | User Manual

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3.4 VoIP

3.4.1 SIP Config

Setting page of public SIP server:

Username of your SIP account (Always the same as

the phone number)

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 IP100 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.IP100

won’t send register info to SIP server if disable register.

Enable Message Waiting:The configuration allows/

forbids Message Waiting.

Advanced(SIP(Setting

seconds. IP100 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.

No 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, IP100 will periodically

detect if the SIP server is available 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: The particular service 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 IP100 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 Iax2 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.

IP100 VoIP Phone | User Manual

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

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 letters 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

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. Network device

connecting to the IP100 LAN port can dynamic obtain

the IP in the range between start IP and end IP.

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.

User may use below setting to add a new lease table.

Notice: This setting won’t take effect unless you save

the config and reboot the device

3.5.2 NAT

Advance NAT setting. Maximum 10 items for TCP and

UDP port mapping.

DHCP Lease Table：Show IP—MAC corresponding table

assigned by DHCP server.

IPSec ALG: Enable/Disable IPSec ALG;

FTP 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.

Click Add to add new port mapping item and Delete to

delete current port mapping item.

IP100 VoIP Phone | User Manual

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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. When 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.

3.5.5 Firewall settings

Firewall setting page：User may set up firewall to

prevent unauthorized 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.

IP100 firewall filter is base WAN port. So the source

address or input destination address should be WAN

port IP address.

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