Telematrix 3300IP-TRM User Manual

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3300IP-TRM VoIP Phone

User Manual

TeleMatrix, Inc

1 Introduction

............................................................................................................5

1.1 Overview of Hardware......................................................................................5

1.1 Overview of Software

2 Keypad of 3300IP-TRM

2.1 Function Table of Keyboard

2.2 Keyboard function and designed catalog

3 Set the Phone Through WebBrowser

3.1 Login

................................................................................................................10

........................................................................................5

.........................................................................................6

.............................................................................6

........................................................9

...............................................................10

3.2 Current state....................................................................................................10

3.3 Network............................................................................................................11

3.3.1

Wan Config

Lan Config.................................................................................................13

3.3.2

3.4 VoIP

3.4.1

.................................................................................................................13

SIP Config

Iax2 Config................................................................................................15

3.5 Advance

...........................................................................................................16

...............................................................................................11

.................................................................................................13

DHCP Server............................................................................................16

3.5.1

NAT...........................................................................................................17

3.5.2

3.5.3

3.5.4

3.5.5

3.5.6

3.5.7

3.5.8

3.5.9

3.5.10

3.5.11

3.5.12

.........................................................................................................18

STUN

Net Service...............................................................................................19

Firewall settings

........................................................................................20

VLAN Configuration

Digital Map................................................................................................21

Call Service Settings

Memory Key..............................................................................................24

MMI Filter................................................................................................24

Audio Settings

.........................................................................................24

VPN.........................................................................................................25

3.6 Dial-Peer dial rule setting

3.7 Config Manage

................................................................................................27

..................................................................................21

................................................................................23

...............................................................................26

3.8 Update Firmware ............................................................................................28

3.8.1

Update

Auto Update..............................................................................................29

3.8.2

3.9 System Manage

3.9.1

Account Manage

Syslog Config............................................................................................30

3.9.2

3.9.3

Phone Book

Time Set....................................................................................................31

3.9.4

MMI SET...................................................................................................32

3.9.5

3.9.6

Logout & Reboot

.......................................................................................................28

..............................................................................................30

.......................................................................................30

..............................................................................................31

.......................................................................................32

4 Operating Method for Dialing

4.1 How to dial IP Phone

4.2 Set the Phone to Server

Set WAN Interface....................................................................................33

4.2.1

4.2.2

SIP Setting

......................................................................................33

.................................................................................33

................................................................................................35

............................................................................33

IAX Setting................................................................................................36

4.2.3

4.3 How to Use Dialing Rules...............................................................................36

4.4 Voice mail

........................................................................................................38

If phone’s default IP address has changed from 192.168.1.179, the current IP address can be displayed by using the submerged keys below the faceplate.

Under on-hook status, press “Down” for seeing IP address:

After reboot, it will to gain the FTFP Server address through the DHCP Server, then "Enter ConfigID" will be displayed on the screen. After input the ID by numeric keyboard, input ”#”, then the phone will download configuration files from the TFTP Server automatically, and if succeed, it will auto-reboot; if failed, it will be into the standby condition and can log on ; if do not want to download, you can press # into the standby condition directly. If the downloading hadn't been finished or the config file name in the autoupte module of the configuration files downloaded with no configuration parameters, after reboot, the "Enter ConfigID" will still be displayed on the screen.

Function

Support two SIP server working at the same time

Provide a Backup SIP Server

Support NAT, Firewall

4. Support DHCP assign IP address, etc automatically

5. Support PPPoE（used while connecting ADSL，cable modem）

It can update the program through HTTP ,FTP and TFTP

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

Hold Function

9. Hotline Function

10.

Speed-dial

11.

Call-forward, Three-way conference call

12. Caller ID display

13.

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

14. Auto-answer.

15.

Set through standard Web Browser

16.

Remote Management Function

17. Classification management for common user’s password and

superuser’s password.

Standard 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  RTP: Real-time Transport Protocol  RTCP: Real-time Control Protocol  VAD/CNG save bandwidth  Telnet: Internet's remote login protocol  DNS: Domain Name Server  TFTP: Trivial File Transfer Protocol

1. Introduction

This is the user manual of 3300IP-TRM. Some configuration should be done before use the 3300IP-TRM phone, then it can work normally. This manual will illustrate how to set the phone through keyboard and web service.

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,

network.

password“admin”to login and set.

1.1.2

Only the WAN interface support the POE.

1.2 Overview of Software

Lan is web-bridge mode, and bridged the LAN and WAN into the same

. You can use the administrator’s user name “admin” and

Network Protocol Tone

 SIP v1(RFC2543)

V2(RFC3261)

 IP/TCP/UDP/RTP/RTCP  IP/ICMP/ARP/RARP/SNTP

 Ring Tone  Ring Back Tone  Dial Tone  Busy Tone

 TFTP Client/DHCP Client/PPPOE

Client

 Telnet/HTTP Server  DNS Clients

Codec

 G.711： 64K bit/s(PCM)

Phone Function

 V olume Adjustment  Speed dial key

 Phonebook  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

IP Assignment

 IP (Static IP)

 DHCP

 PPPoE

G.729

Voice Quality Security

 VAD： Voice activity detection  CNG： Comfortable noise

generator

 LEC： Line echo canceller  Packet Loss Compensatio n  Adaptive Jitter Buffer

 HTTP 1.1 basic/digest

authentication for Web setup

 MD5 for SIP authentication

(RFC2069/RFC2617)

QoS

 QoS field

Call Function NAT Traversal

 STUN  Call Hold  Call Waiting

Configuration

 Call Forward

234

678

910

 Caller ID  3-way conference

 Web Browser

 Console/Telnet

 Keypad

DTMF

 DTMF RELAY  DTMF RFC 2833  DTMF SIP Info

SIP Server

 Support two SI P serv er working at

the same time

 Provide a Backup SIP Server

2 Handset key of 3300IP-TRM

Firmware Upgrade

TFTP



HTTP



 FTP

1、Softkey1 key 2、Softkey2 key 3、Softkey3 key 4、LCD Screen 5、Dial Pad 6、SEND & # key 7、Redial key 8、Up key 9、Down Key 10、Hold key

2.1 Function table of keystroke

Name Status Function/Display

Softkey1

Softkey2

Correspond to the function on the lower left side

of the screen according to different states.

Correspond to the function on the lower middle of

the screen according to different states.

Softkey3 Correspond to the function of the lower right side

of the screen according to different states.

Redial

Dialing Redial the number dialed last time On-hook Check the call record On-hook

Check the registration of SIP1、SIP2、IAX2

Call Increase the volume

Config Roll up/left the item bar

On-hook Check the IP address of the phone

Down

Call Decrease the volume

Config Roll down/right the item bar

Dial Pause

Hold

Call hold

On-hook Do not disturb

Dialing “1”

Config “1”, “@”, “/”, “:”, “+”, “-”, “%” “!”, “|”,

Dialing “2”

Config “2”, “a”, “b”, “c”, “A”, “B”, “C”

Dialing “3”

Config “3”, “d”, “e”, ”f”, “D”, “E”, “F”

Dialing “4”

Config “4”, “g”, “h”, “I”, “G”, “H”, “I”

Dialing “5”

Config “5”, “j”, “k”, “l”, “J”, “K”, “L”

Dialing “6”

Config “6”, “m”, “n”, “o”, “M”, “N”, “O”

Dialing “7”

Config “7”, “p”, “q”, “r”, “s”, “P”, “Q”, “R”, ‘S”

Dialing “8”

Config “8”, “t”, “u”, “v”, “T”, “U”, “V”

Dialing “9”

Config “9”, “w”, “x”, “y”, “z”, “W”, “X”, “Y”, “Z”

Dialing “0”

Config “0”, “space”

Dialing “*”, “#”, “@”, “,”, “.”, “: ”, “/”, “$”, “%”, “&”, “(”, “)”, “<”,

“>”, “[”, “]”, “~”

Config “*”, “.”

# Dialing It can be regarded as the first number being

dialed out or the end mark for ending number.

2.1.1 Voice Control Press “Up”to increase the volume and press “Down”to decrease.

2.1.2 Hold Function Hold the current call line.

2.1.3 Redial Fuction

Redial the number dialed last time.

2.1.4 Pause Function Press "Hold" key while dialing, it will display "^" on the screen to show that

the phone should send the numbers behi nd "^" after being connected to the receiver for 3.6 seconds.

Attention:

2.1.5

Three-way conference call

Pause can not use as the first number.

Suppose the user of the 3300IP-TRM 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 convers ation 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 Do not disturb function

1、Enable the Do Not Disturb function:

Press "Hold" key under the On-Hook st atus , then the ph one wil l enabl e the function of Do Not Disturb, and it will keep displaying "Do Not Disturb" on the screen on that status. 2、Disable the Do Not Disturb function:

Press "Hold" key under the On-Hook status, and the phone w il l dis pl ay

" Allow Calls" on the screen, then back to the normal standby screen.

2.1.7 Call Record Press "Redial" key to enter Call Record menu. 1、 Outgoing Call

Check the Outgoing Call record. If yo u have made a outgoing call, that press "Redial" key to check the outgoing record directly under the On-Hook status, now if you need to check the "Incoming Call" and " Missed Call ", press "Quit" key back to the main menu. 2、 Incoming Call

Check the Incoming Call Record.

3、 Missed Call

Check the Missed Call record. If the phone has missed calls, that press "Enter" key to check the missed calls directly on the status of On-Hook, now if need to check the "outgoing call " and "Incoming Call", press "Quit" key back to the main menu. 4、Methods for Number details

Find the number by checking the call record, and then press "Detail" for the number details, and the details can be checked with the keys "Up" and "Down". 5、Callback method

Find the number by checking the call record, and press "Detail" for the number details, then press "Dial". Now, if you want to make an outgoing call but the line is on internal net, it only needs to input the out group number, then press "Dial" key so that the out group number will be added before the number you checked.

2.2 Keyboard functions and setting catalog

1、Press down the "Menu" key to enter the menu under the ON-Hook status. 2、It needs password to enter the Advanced menu, and the default password of the phone is 123.

3、Menu catalog:

1) Screen Set

2) Ringer Set

3) Volume

4) Greeting Word

5) Call Service

6) Advanced

7) Reboot System

2.2.1 Screen Set：

Contrast

2) Brightness

2.2.2 Ringer Set

Ringer Volume

2) Ringer Type

2.2.3 Volume

1) Voice Volume

2) Mic Volume

2.2.4 Greeting Word

2.2.5 Call Service

Do Not Disturb

a) Mode

 OFF  Always  Busy  No Answer

b) Number

2) Call Forward

Call Waiting

4) HotLine

5) Dial Rule

a) End With #

2.2.6 Advanced

1) Set Password

SIP Set

Network

2.2.7 Reboot System

b) Fixed Length

 Switch  Value

c) Time Out

a) SIP Server b) SIP Number c) SIP Account d) SIP Password e) SIP Register

a) Net Mode

 Static  DHCP  PPPoE

b) Static Set

 IP  Netmask  Gateway  DNS

3 Through web browser to set 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 s et the comput er IP on the same net with t he phone IP or make it gets IP automatically . Then open the IE, input the phone IP address in the address field, t you will enter web setting page of phone 3300IP-TRM.

3.1 Login：

The default user name and password are admin/admin and guest/guest.

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 telep hone 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 3300IP-TRM’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, 3300IP-TRM 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.

3300IP-TRM will automatically obt ain 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: 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: Bridge Mode:

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 NAT.

Set the IP and Netmask for the LAN

Enable this option to switch to bridge mode. IP phone won’t assign IP

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)

Password of your SIP 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 Password:

proxy server username

proxy server password

Domain Realm: SIP domain, enter the sip domain if any, otherwise 3300IP-TRM will use the proxy server address as sip domain.

Local SIP port: Local SIP register port, default 5060

Phone Number:

Enable Register:

Phone number of your SIP account

Enable/Disable SIP register.3300IP-TRM won’t send

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

Advanced SIP Setting

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

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.

for detail.

Detect Interval Time:

enable, 3300IP-TRM 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: DTMF signal sending mode: support RFC2833, DTMF_RELAY (inband audio) and SIP info

RFC Protocol Edition:

gate need to communicate to devices (such as CISCO5300) using the SIP 1.0. Default is RFC 3261.

Co-work with the Auto Detect Server, if Auto Detect Server is

Current 3300IP-TRM SIP version. Set to RFC 2543 if the

3.4.2 Iax2 Config

Setting page of public IAX server:

IAX Server Addr:

IAX Server Port: Account Name:

number)

Account Password: Local port: Phone Number:

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, the n you use the number to stand for your username

Voice mail text:

here. Echo test number: If the platform su ppor t e cho te st, a nd t he number i s te st form , th e 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: Refresh time: Enable Register: enable or disable register Enable G.729: IAX2(Default Protocol):

choose SIP as default

IAX refresh time

Using G.729 speech coding ma ndatory consultations

Password of your IAX account.

Signal port of local, default port is 4569

if IAX support voice mail, config the domain name of your mail box

echo test number in text format

Phone number of your IAX account

Set IAX 2 as the default protocol , if not the system will

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:

Start IP:

End IP: End IP of lease table. Network device connecting to the 3300IP-TRM LAN

port can dynamic obtain the IP in the range between start IP and end IP.

Lease Time:

Netmask:

Gateway:

DNS: default DNS server of lease table.

DNS Relay:

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

Start IP of lease table.

Netmask of lease table.

Default gateway of lease table

enable DNS relay function.

Notice: This setting won’t take ef fect unless you save the config and reboot the device

Lease table name.

DHCP server lease time.

3.5.2 NAT

Advance NAT setting. Maximum 10 items for TCP and UDP port mapping.

DHCP Lease Table

IPSec ALG:

FTP ALG:

PPTP ALG:

Transfer Type:

Enable/Disable IPSec ALG;

Enable/Disable FTP ALG;

Enable/Disable PPTP ALG;

Transfer type using port mapping.

Inside IP: LAN device IP for port mapping.

Inside Port:

Outside Port:

LAN device port for port mapping.

WAN port for port mapping.

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

：Show IP—MAC corresponding t able assi gned by D HCP ser ver.

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 STU N is u sed to real iz e 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: STUN Server Port: STUN Effect Time: stun detect NAT type circle, unit: minute.

configure stun server address;

configure stun server port default 3478

Local SIP Port： Load： Use Stun

Load the choices of SIP line.

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

The SIP port of this phone.

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 Port Quantity: Maximum RTP port quantity, default is 200

RTP initial port.

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 support 10 items maximum.

3300IP-TRM firewall filter is base WAN port. So the source address or input destination address should be WAN port IP address.

Configuration:

In_access enable

Out_access enable enable out_access rule

Input/Output:

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

Protocol Type:

Port Range:

port range if this rule

Src Addr: source address. Can be single IP address or network address.

Dest Addr:

destination address. 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

enable in_access rule

specify current adding rule is input rule or output rule.

protocol using in this rule: TCP/IP/ICMP/UDP.

3.5.6 VLAN Configuration

3300IP-TRM phone implement QoS based on 802.1p, The QoS is used to mark the network communication priority in the data link/MAC sub-layer. 3300IP-TRM will sort the packets using the QoS and sends it to the destination.

VLAN Enable: If enable the VLAN service, the second layer will realize separate voice, signal and data transmission. To realize separate voice and data transmission by dispose for IP precedence of ToS area of voice transmission. To reach upper layer switch or router have priority to transfer voice transmission. (The prerequisite is the upper layer switch or router has to identify ToS area.)

VLAN ID:

function. The realized voice packet s tra nsfer at the s ame VLAN. The prerequ isite is it must the same as VLAN of upper switch. The value range are 1~4094.

DiffServ Enable: If enable the VLAN service, i t indicates use DSCP mode to realiz e three layers QoS. This moment, the DSCP of SIP signals which between 3300IP-TRM 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 the values of DiffServ Value field.

Dispose VLAN ID is add a Tag header after realize enable the VLAN

DiffServ Value:

0x28,0x30,0x38,0x48,0x50,0x58,0x68,0x70,0x78,0x88,0x90,0x98,0xb8.default is 0xb8 ,oxb8 stands for best fast transmission; 28-38 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 transmi ssi on pri ority f or the 3r d r ank, 88- 98 i s guar antee f or t he transmission priority for the 4th rank.

802．IP Priority: The priority of 802.ip

The value range：

3.5.7 Digital Map

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

3300IP-TRM support below digital map:

Digital Map is based on some rules to judg e when user e nd thei r dialin g an d se nd the number to the server. 3300IP-TRM support 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. The call will be sent after timeout

----Prefix: User define digital 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 represents any one digit between 0~9

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

Example:

8[2-8]xxx xx All number from 8200000 to 8899999 will be sent immediately. 955xx 5 digits numbers begin with 9 will be sent immediately. 10060 Number 10060 will be sent will be immediately 22xxxxxT1 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.

3.5.8 Call Service Settings

User 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. 3300IP-TRM immediately dials this number after hook-off if it is set.

Auto Answer: Enable/disable auto answer function.

No Disturb:

Ban Outgoing:

Enable Call Transfer: Please refer to Value_add_service

Enable Call Waiting:

Enable Three Way Call:

Accept Any Call: If this option is disable, 3300IP-TRM refuse the incoming call when

the called number is different from 3300IP-TRM’s phone number.

No Answer Time: no answer call forward time setting.

Black List:

Limit List:

DND, do not disturb, enable this option to refuse any calls.

Enable this to ban outgoing calls.

for detail.

Enable/disable Call Waiting

Please refer to Value_add_service

for detail.

incoming call in these phone numbers will be refused.

o utgoing calls with these phone numbers will be refused

3.5.9 Memory Key

This page layout shows number setting of Voice mail.

3.5.10

When MMI filter is enable. Only IP address within the start IP and end IP can access 3300IP-TRM phone.

MMI Filter

MMI filter is used to make access limit to 3300IP-TRM phone.

3.5.11 Audio Settings

CODEC:

Signal Standard:

Handdown Time: hand down detect time.

Input Volume:

Output Volume:

Handfree Volume: Hand free volume

G729 Payload Length:

VAD: Enable/disable Voice Activity Detection

select the prefer CODEC; support ulaw, alaw, G729 and G7231 5.3/6.3

Handset in volume.

Signal standard for different area.

Handset out volume.

G729 payload length

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：

After VPN registered successfully, VPN server will give an IP aggress to the

terminal. If there is a IP addr ess shown on ter minal ( except for 0.0. 0.0),i t means your VPN has registered

UDP Tunnel

VPN Server Addr：

VPN Server Port

Server Group ID

： Register to the port of VPN server

： the group ID of UDP VPN

Server Area Code： the area code of VPN server

L2TP

VPN Server Addr： register to the address of VPN server

VPN User Name： L2TP VPN username

VPN Password：

UDPTunnel:

L2TP VPN password

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.

Dial-Peer dial rule setting

3.6

Please refer to How to use dial rule

for detail.

Config Manage

3.7

Save Config: save current settings. Clear Config:

restore to default settings.

Backup Config: Backup the config file, via point the right key of mouse- save target as….-will pop a save window, then type the confi g file na me in t h e F ile 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.

3.8 Update Firmware

3.8.1 Update

Web Update：

Update the application or configuration files of the phone. The application document is .z format, and the configuration files is .cfg format.

Through clicking on the "browse" button to open the upgrade file or configuration file, then click on "Update" button. After the upgrade, 3300IP-TRM 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

Config file export: export the config file

Config file import: import the config file

3300IP-TRM phone support FTP and TFTP auto update, t he gateway will auto obtain the configure file from your update server if configur ed. To obtain the original config ure file, you can use the FTP/TFTP back up as describe above. Configure 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.

3.8.2 Auto Update

Current Version:

Server Address:

Username: FTP server user name

Password:

Config File Name:

Config Encrypt Key: The encrypt key of confirmation file

Protocol Type:

Update Interval Time: The interval time that the terminals search for new

configuration file.

the system will display the current version number

FTP/TFTP server address

FTP server password

The name of configuration file

The protocol type that used for upgrading

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

Configure 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>>Ve rsion:1.0000

<GLOBLE CONFIG MODULE> ConfFile Version：6

User may edit the configure file version to:

<<VOIP CONFIG FILE>>Ve rsion:1.0007

<GLOBLE CONFIG MODULE> ConfFile Version：7

3.9 System Manage

3.9.1 Account Manage

Set web access account or keypad password of 3300IP-TRM.

3.9.2 Syslog config

Set the system log

Server IP:

Server Port: set the syslog server port

MGR Log Level:

SIP Log Level:

IAX2 Log Level: set the IAX2 log level

Please click “apply” after setting

set the syslog server address

set the MGR log level

set the SIP log level

3.9.3

Phone Book

3.9.4 Time Set

This page layout is the setting of time of phone.

Server:

Timezone: select correct time zone in list box

Timeout:

Daylight:

SNTP: select SNTP server

12 Hours Format:

Manual Config: The time setting

type the IP address of time server

longest response time for SNTP

Daylight saving time

select 12 hours format

3.9.5 MMI SET

Set the greeting information on LCD.

。

3.9.6 Logout & Reboot

Logout：logout the Web entry.

Reboot Phone：logout the entry, and reboot the phone. When user modify any config of the phone, it will take effect after being rebooted, you can enter into this layout and click

“Reboot”. And the phone will be rebooted automatically.

Note：

always save config before reboot, otherwise the setting will return to previous setting.

Reboot IP phone, some setting needs to reboot to make it works. Please

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. Please confirm whether all the net wires are connected correctly.

If you want to make a call, you can ma ke it af ter di aling th e num ber and th en pressi ng “#”.

You can find IP address by the menu.

Modifying the IP address of the computer, and making it the same net with IP100.

Inputting the IP address of IP100 in the browser, and then you can visit the setting layout of IP100 after press the Enter key; super user account is admin/admin; common user account is guest/guest.

4.2 Set the phone being connected to server

4.2.1 Set the WAN interface

The connection ways of entering the NetworkWAN Config layout phone of the net port:

3300IP-TRM could be connected to Internet by using the static IP, DHCP IP, or PPPoE dialing.

Configure Static IP:

----choose static;

----fill in the IP address of 3300IP-TRM 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 address in the Primary DNS and Alter DNS respecti vely.

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

----choose DHCP option. Now, if the network has DHCP server, then 3300IP-TRM will get IP address, Netmask, Gateway, Primary DNS and Alter DNS from this DHCP server automatically.

Use PPPoE dialing to connect the Internet:

----choose PPPoE option.

----please fill in the account and password which PPPoE have dialed in the PPPoE Username and Password. So 3300IP-TRM could connect the I nt ern et t hr ou gh PPPoE dialing, and automatically get IP address, Netmask, Gateway, Primary DNS and Alter DNS and so on .

4.2.2 SIP

setting：

Enter into the information:

phone number)

VoIP  SIP Config

to set the layout config and sip account

Username of your SIP account (Always the same as the

，default port is 5060

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 .

Password of your SIP account.

4.2.3 IAX setting

IAX Server Addr:

IAX Server Port:

Account Name:

number)

Account Password: Local port: Phone Number:

----choose Enable Register；

----if you use IAX account to m ake a call, please choose IAX（Deault 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 are being set correctly .

Note:

please choose Save Config in the Config Manage after setting the information, or

the existing setting information will be failed after rebooting..

How to use the dial rule?

4.3

3300IP-TRM provide flexible dial rule, with different dial-rule configure, user can

Register address of public IAX server Register port of public IAX server，default port is 4569 Username of your SIP account (Always the same as the phone

Password of your IAX account.

Signal port of local, default port is 4569

Phone number of your IAX account

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 “Add” to add a ne w 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 t his di al-r ule. 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 phone 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 need to

co-work with the Del Length:

add:xxx, add prefix to the phone number, can set to reduce the dial length. 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 Length

rep:xxx，replace the first N numbers. N is set in the Del Length. For Example: Use wants to place a 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.

Suffix (optional):

Configure suffix, show no suffix if not set

Instance:

2T rule

sent to private SIP server.

3T rule: If the call starts with 3, the first 3 will be deleted, 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

dial 06332221015 and AG-188 will send 866332221015 to your server.

11 rul e

without set up a sip server.

: If the call starts with 2, the first 2 will be deleted, and the rest number will be

: If the calls is begin with 0, the first 0 will be replace by 86. Means that if you

: when you dial 11, the call with send to 192.168.0.11, suit for LAN application

4.4 Voice mail

When there is a mail, voice mail LED would be flickering, and LC D would display “New message”. After listening the message, voice mail LED would stop flickering, and “New message” would disappear from LCD.

Telematrix 3300IP-TRM User Manual

Specifications and Main Features

Frequently Asked Questions

User Manual