This Manual provides basic information on how to install and connect IPH303 IP
Phone to the network. It also includes features and functions of IPH303 IP phone
components, and how to use them.
15.3 Structure of Configuration terminal....................................................60
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Thank you for your purchase of our products ---- CDX-IPH303 enterprise-class
IP phones. You make an excellent choice; we hope you will fully enjoy all the
features of the product.
CDX-IPH303 is a highly innovative enterprise IP telephone, to provide you the
plenty of features and beautiful sound quality. Products fully comply with the
SIP protocol standard, and Interconnection with the majority of SIP
interoperability of hardware and software equipment of the market.
The content of this manual is subject to change without prior notice.
Safety Instructions
1
Warning:
Read the safety precautions and the user guide before use.
Explain their contents and the potential hazard associated with using the
telephone to your children.
Liquids of any kind
Don’t expose your phone to water, rain, extreme humidity, sweat,
or other moisture.
Dust and dirt
Don’t expose your phone to dust, dirt, sand, food, or other
inappropriate materials.
Extreme heat or cold
Avoid temperatures below -10°C/14°F or above 45°C/113°F.
Cleaning solutions
To clean your phone, use only a dry soft cloth.
Don’t use alcohol or other cleaning solutions.
Microwaves
Don’t try to dry your phone in a microwave oven.
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The ground
Don’t drop your phone.
Safety Standards
CDX-IPH303 follows various safety standards, including FCC / CE. The
power adapter of product follows the UL standard, the phone can only be
used the power adaptor provided by inner packaging, the damage caused
due to the use of other power adapter, which does not belong to the scope
of quality assurance of the manufacturer.
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Before Getting Started
2
Before you can connect CDX-IPH303 to the network and use it, you must
have a high-speed Internet connection installed. A high-speed connection
includes such environments as DSL, cable modem, and a leased line.
CDX-IPH303 IP phone is a stand-alone device, which requires no PC to
make Internet calls. CDX-IPH303 IP is fully compatible with SIP and IAX2
industry standard and can interoperate with many other SIP or IAX2
compliant devices and software in market.
3 Package contents
1. CDX-IPH303 IP phone base unit.
2. Handset
3. Coiled handset connecting cord
4. One Straight Ethernet cable
5. One Power supply
6. User guide
7. 2 wall mount bracket
8. 1 table bracket
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4 SPEC and Features
4.1 Hardware Spec
l 32-bit 150MHz MIPS CPU
l 16-bit 100MHz DSP
l 4MB flash memory
l 16MB SDARM
l 128x64 pixel graphic LCD with backlight
l 34 KEYS, 1 LED
l WAN: 10/100M RJ45
l LAN: 10/100M RJ45
l RJ9 Headset Jack(option)
l 2.5MM earphone Jack(option)
4.2 DSP Spec
l G.711A/u, G.723.1(5.3k/6.3k), G.729a/b,G.722,G726
l G.168 echo cancel
l Full duplex hand-free
l VAD (Voice Activity Detection)
l CNG (Comfortable Noise Generation)
l AEC (Acoustic Echo Cancellation)
l Adaptive voice jitter buffer
l Codec negotiation supported for fixed and dynamic
4.3 Software feature
l Languages: English
l RFC Protocol Edition: RFC3261 and RFC2543
l Compatible with all major SIP Servers: Cisco, Osip, Vocal, ser, Partysip,
Simens, Grandstream, etc.
l Server authentication mode: none, basic, MD5
l Peer to Peer SIP call
l Line 1/ Line 2. can support two different SIP servers.
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l DTMF Mode: RFC2833, RELAY, SIP INFO
l DNS name of SIP server
l SIP signaling port setting
l NAT traverse, STUN
l NAT traverse, SIP Express router
l Flexible Dial Map: Fix length; End with #; Dial with time out
l 9 Kinds of ringer able select by number of Phone Box and 2 kinds of
ringer user defined
l Speakerphone
l Headset
l Dial Map Table
l 5 Speed dial key.
l Black list for reject authenticated call
l Reject incoming call
l Limit dialing out No. list
l No Disturb
l Caller ID display
l Call forward, call transfer, call hold, call waiting
l Call forward with unconditional, busy and no answer
l 3 party conference
l 50 entries each for dialed call, received call and missed call
4.5 Networking Standards
l WAN/LAN port with Router or Bridge Mode
l NAT ALG
l PPPoE for xDSL, automatically keep alive
l DHCP Client on WAN
l DHCP server on LAN
l DNS client with 2 servers IP
l SNTP
l RTP: RFC3550
l 802.1P QOS
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4.6 Others
l Boot Monitor
l Upgrade firmware through POST mode
l Keyboard Configuration
l HTTP Web-Based Configuration
l FTP, TFTP upgrade firmware·
l WEB upgrade firmware
l FTP, TFTP, HTTP upload/download configuration file
4.7 Physical & Environmental
Desktop / Wall mounting
Power Input: 100 to 240 ACV / 50/60Hz Output:7.5 DCV
Dimensions: 210 x 170 x 130 mm
Weight: 680 g (main unit)
Operating Temperature: 0°to +40°C (32°to +104°F)
Storage Temperature:-20°to +70°C (-40°to +158°F)
Humidity: 5% -95% non-condensing
5 Installation
Use the cable for Internet connection; connect the ‘WAN’ port of
CDX-IPH303 to router. Get the cable from box and connect the LAN port of
phone to your PC. Connect the power adapter in the box to ‘DC’. Then start
your phone.
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Power Jack 7.5V DC power connected port
LAN
WAN
10/100Mbps RJ-45 PC connected
port
10/100Mbps RJ-45 Ethernet
connected port
RJ9 Headset Jack
Handset Jack
Desktop installation:
Wall installation
This phone can be installed on the wall, the back of base has two wall
-mounted ports:
To install the phone on the wall, put two pothooks on the wall, hang the wall
port of the phone back on the wall. The two supporting frames will balance
the bottom of the phone and place it to the wall
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Date and
Time
OFF ON
Use the handset, pull out the wall-mounted label (extension of the down)
from the handset bracket, rotate the wall label and then insert to
wall-mounted label slot, caught by the extension of the handset.
6 General Appearance
6.1 LCD Display
Non register
Registered
IP address
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1
LCD illustration Definitions
Call forwarding enabled
2
Call on hold
3
Connected call
4
5
6
7
8
9
10
11
12
13
Incoming call
Conference call active
Conference call on hold
Conference call disabled
Outgoing call
Outgoing call not completed
Transferring a call
Line 1 (idle)
Line 2 (idle)
Line disabled
14
Handset in use
15
Speakerphone in use
13
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Message waiting
17
6.2 Key function Definitions
Missed call
Key Button Key Button Definitions
1 0 - 9, *, #
Digital, star and pound keys are also used for setting
and call process.
2
3
Soft key 1、2、3 , Display functions as per the LCD
menu requested
Menu: enter the main menu, or confirm current status
14
4
5
6
7
8
9
10
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LINE1 and LINE2 : It is the transparent key, it will be
flash when incoming call, it will be normal on the call, it
will be flash on Hold.
Hold: hold the active call.
The transparent key has red indicator.
Call List: browse the call logs
Mute key: Used as Mute and quiet on the call, The transparent
key has red indicator.
Phone Book: Enter to the phonebook to recall and
amend the phone numbers.
Navigation key used as UP in menu or Increase
handset / speakerphone volume
Navigation key used as DOWM in menu or reduce
handset / speakerphone volume
Navigation key used as LEFT in menu or reduce
handset/speaker volume
12
13
Navigation key used as RIGHT in menu or increase
handset/speaker volume
Headset: activate/deactivate the headset function
14
15
Speaker: activate/deactivate the hands-free function
Redial: Dial a new number or redial the last call.
16 M1-M5
Speed dial: make the speed dial call
6.3 Indicator Instruction
Key light Definitions
1
For each line:l On (steady)-Active call
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l Slow blink-Held call
2
l Fast blink-Incoming call
On – Speaker in use
3
On – headset in use
4
On – Phone is muted
5
On – Phone is held
6 M1~M5
LCD
7
Backlight
BLF indicator status:
l ON:appointed extension is being used
l OFF:appointed extension is on line and on standby
FLASH:appointed extension is off line and not registered
Backlight status:
1. Always light on standby : missed call or voice
message.
2. Always light on using.
7 Phone Operations
7.1 Point to Point Call
Pick up or press the 【】, 【 】 key,【 】、【】,then input “#Phone
IP address#” for example:The IP phone of the other side is:192.168.0.11,
it should be when using keyboard input:#192*168*0*11**5060#,* indicates
point<.>,** indicates colon<:>
7.2 Two-SIP account and line
CDX-IPH303 support 2 independent SIP accounts. Each account can
support independent SIP server, user name and NAT configuration. Line
button 【 】、【 】 are match to sole SIP account , when picking, press
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a free line (【 】 or【 】) button, and at the same time hear the dial tone.
In this state, use UP / Down keys can be cut between the two lines.
7.3 Make a call
1. Use headset, press <SPEAKER> or LINE1/LINE2 key
Step 1: pick up / press 【】 key / use a 【 】 or press 【 】or 【 】
key (activated).
Step 2: Telephone dial tone will be sounded and 【 】 LED indicator will be
light. You can press 【 / 】 key to select the other SIP accounts
Step 3: input the number, press 【#】 key to send
2. Use the redial key
Redial the last called number (redial, the phone will use the SIP account of
the last call)
Step 1: pick up /press 【】 key / use a 【 】 or 【 or 】 key,
the corresponding LED will light and hear the dial tone (The function is not
available now and will be finished some time).
Step 2: Press 【】Redial key to dial the last dialed number.
Check the dialed record and redial
Step 1: On the standby, press 【】 key to check the dialed record,
Step 2: Use the 【 / 】 key to select the corresponding call records,
Step 3: Press Redial/pick up/Speaker/LINEx/Headset.
3. Use the phonebook menu to make the call
Press Phone book key to enter the phonebook menu, then select the dialed
number, as per the prompt of soft key to process. Use the related SIP
account to make a call with hand free.
4. Use the call record to make the call
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Press Soft Key 2 to enter the Calllog menu,use 【 / 】 key to check
the Missed Call/Incoming Call/Outgoing Call record, select the dialed
number, then pick up or press 【】【 】 【 or 】 to make the
call.
5. Standby dial-up call
On standby, dial the called number ,press the soft key 3<Dial>or pick up,
press【】 or【】【 or 】,only use LINE 2 key to call
from the second line, the other defaults are from LINE 1.
6. Use the phonebook number to make the call
On standby, press the Phone book key to enter the phone booklist , select
the dialed number, as per the prompt of Soft Key to process the operation,
you can pick up, press 【】or【】【 or 】 key to call.
7.4 Phone book number store / edit / delete
On standby .press Phone book key, then as per the prompt of Soft Key to
process Store/Edit/Delete.
7.5 Call list check / delete
On standby , press Soft Key 2 to enter Call Log menu,then as per the
prompt of Soft Key to process Check / Delete.
7.6 Hold
During conversation, press Hold to keep line, press it again to release.
7.7 Mute
During conversation, press Mute to keep line, press it again to release
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7.8 Volume adjustment in conversation
During conversation, press navigation 【 / 】 key to adjust receiving
volume, or adjust the volume on menu.
7.9 Rejected call
During new incoming call, as per the prompt of the LCD Soft Key,press soft
Key 3 <REJECT> to reject the incoming call.
7.10 Black(White) list setting / edit / delete
Please refer to below setting menu and CALL SERVICE setting in web
configuration.
7.11 Call transfer
During conversation,as per the prompt of LCD Soft key,press Soft Key
3<transfer>, and input transferred number end with 【#】 to transfer the
phone to the third part and hang up automatically
7.12 3-party conference
During conversation,as per the prompt of LCD Soft key,press Soft Key
2<Conf>, then dial another number plus # button, while line connected,
press Soft Key 2<Conf> to make conference call;
7.13 SMS Function
Ø The function standard RFC3428(Session Initiation Protocol (SIP)
Extension for Instant Messaging)
Ø This feature must need the support of the SIP server and end phone
Ø Function operation
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7.13.1 Create/New . send message
1. On standby state, press Soft Key1 <SMS>to enter SMS menu , see
below photo
Inbox x/x:The first letter X means the missed message number, and
last letter X means the total message number.
New: Create the new message,Quit: Back to SMS menu
2. New/ Create the new message,see below picture:
3.Send message: input the extension number and press soft key 2 to
send message or select the number on notebook and press soft key 2 to
send message.
7.13.2 Read, delete and reply message
1. On standby , press Soft Key <SMS> to enter to SMS menu,as below
picture:
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2. Press Soft Key 2<Enter>,you can delete or enter to the message as
per the prompt of the Soft Key.
3. Press Soft Key 2 <Enter> to check the messge,you can reply or
delete the message as per the prompt of the Soft Key.
4. Press Soft key 1<Reply> to reply the message,enter to the state to
input the message, after finished inputting, then press Soft key 1<Reply>
to send the replied message.
8 Configuration with keypad and LCD display
8.1 Main menu list operation
In standby, press 【MENU 】button till LCD shows
-Configuration
-Advanced
-Option
Three Sub-menu option,“Configuration”is used the configured property
parameter,configured the advanced parameter and password protected, the
default password is 123,Option the other related functions.
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Use 【】and【】key to select the related sub item ,and process the
operation as per the LCD soft key.
The instruction for the main menu list as below
8.2 Phone book operation
On Standby, press Phone Book to enter the phone book menu list, use
【】and【】key to select the sub item, and process the recall, check, edit,
delete etc. as per the LCD soft key
8.3 Call list check
On standby , press Soft Key 2 to enter Call log menu,use【】and【】
key to select the related sub item, and process check, delete, edit ,save etc
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as per the LCD soft key.
the instruction for call list as below:
8.4 SMS operation
On standby, press Soft Key 1to enter SMS menu,use【】and【】key
to select the related sub item, and process the related operation as per the
LCD soft key.
8.5 Memo operation
On standby, press Soft Key 3 to enter Memo menu,use【】and【】key to select
the related sub item,and process the related operation as per the LCD soft key.
9. WEB Interface Configuration
The IP Phone Web Configuration Menu can be accessed by the following
URI: http://Phone-IP-Address. The default LAN IP address is “192.168.10.1”
and WAN IP address is DHCP.
If the web login port of the phone is configured as non-80 standard port, then
user need to input http://xxx.xxx.xxx.xxx:yyyy, otherwise the web will show
that no server has been found.
9.1 Logon Web
While input correct IP address as above, logon menu pop out as follows:
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There are two level login as:
-Guest account: the default username and password is "guest", user can
have a browse of system.
-Administrator account: the default username and password is "admin", this
user can configure the system.
Note: After inputting username and password, user press carriage return
directly to enter the page.
While successfully login, web shown as follows:
9.2 Current State
On this page user can gather information of each normal parameters, as:
-the network section shows the current WAN, LAN configurations of the
phone, including access way of WAN IP and IP(static state, DHCP, PPPoE),
MAC address, WAN IP address of the phone, LAN IP address of the phone,
opening state of LAN DHCP server.
- The VoIP section shows the current default signaling protocol, and server
parameter; Register server IP of SIP, proxy server IP, whether enables
register, whether has registered on register server, whether enables
outbound proxy, whether enables STUN server.
-The Phone Number section shows corresponding phone number of each
protocol; the version number and date of issue have been shown at the end
of the page.
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9.3 Network Configuration
9.3.1 Wide area network(WAN)
User can view the current network IP linking mode of the system on this
page.
User will be authorized to set the network IP, Gateway and DNS if the
system adopts the static linking mode.
If the system selects DHCP service in the network which is using DHCP
service, IP address will be gained dynamically.
If the system selects PPPOE service in the network which is using the
PPPOE service, then the IP address will be gained by the set PPPOE ISP
internet and password of the account.
Note: If IP address has been modified, the web page will no longer respond
owing to the modification, so new IP address should be input in the address
field now.
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Configuration Explanation:
CDX-IPH303-VoIP User Manual V1.3
Current phone IP, subnet mask, MAC address and current phone IP;
,
Select acquisition way of IP for WAN; This is single option; Configure static
IP parameter for WAN;
Configure static IP address;
Configure subnet mask;
Configure IP address of the phone;
Configure "DNS domain" suffix; if user input "domain" and it can't be
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resolved, then the phone will add and resolve the "domain" after user has
input;
Main DNS server IP address;
The second DNS server IP address;
Configure PPPoE:
Service name, if PPPoE ISP has no special requirement for this name,
generally is the default;
PPPoE account;
PPPoE password;
Configure the parameter and then click "apply" to go into effect.
9.3.2 Local area network (LAN)
User can make local area network (LAN) configuration on this page, when
bridging mode is selected, the local area network (LAN) configuration will no
longer go into effect.
Configuration Explanation:
Use bridge mode(transparent mode):bridge mode will make the phone no
longer set IP address for LAN physical port, LAN and WAN will join in the
same network;
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Configure LAN static IP;
Configure LAN subnet mask;
Enable LAN port DHCP server; after user modify LAN IP, the phone will
automatically modify the adjustment and save the configuration according to
IP and subnet mask team DHCP Lease Table, user need to restart the
phone to make DHCP server configuration go into effect;
Enable NAT.
Service Port configuration
User can set up Telnet, HTTP, RTP port on this page and view DHCP table.
Configuration Explanation:
Configure web browse port, the default is 80 port, if you want to enhance
system safety, you'd better change it into non-80 standard port;
Configure telnet port, the default is 23 port;
Enable RTP initial port configuration. It is dynamic allocation;
Configure the maximum quantity of RTP port. The default is 200;
Leased IP-MAC correspondence table of DHCP;
The configuration on t※his page needs to be saved after modified and will
go into effect after restarting.
If the Telnet, HTTP port will be modified, the port is better to be set as ※
greater than 1024, because less than 1024 port system will save ports.
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Set the HTTP port as 0※,then the http service will be disabled.
10 VOIP Configuration
10.1 IAX2 Configuration
Configuration Explanation: Explanation:
IAX2 registration state display; If register successfully, it will display
[Registered], otherwise will display [Unregistered];
Config IAX2 the server address, also can use domain name form;
Config IAX2 server port;
Config IAX2 account name;
Config IAX2 account password;
Config IAX2 phone number;
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Config equipment iax2 monitor port;
Config voice mail number,If the IAX2 support voice mailbox, the voice
mailbox is the letter form, the gateway is unable to input the letter, uses this
number to replace voice mail the name;
Config voice mailbox name; if the IAX2 support voice mailbox, here to config
the vocie mailbox the name;
Config whether supports echo. If the platform support echo, (echo number is
the text format), then the telephone config this echo test number replace
echo actual text number. This function is refers through the platform, the
terminal may carry on echo the call to test. To see the terminal to the
platform converses on the telephone whether normally;
Config echo test text;
Config IAX2 refresh time,The unit of time for the second, suggested the user
in makes the choice 60 to 3600 between;
config the permission/prohibition registers the server;
Config whether supports G.729;
10.2 SIP configuration
Sip register
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Configuration Explanation:
Select SIP1 or SIP2,then you can register and configure SIP1 or SIP2;
Show SIP register state; if register successfully, there will show Registered
in the square bracket, otherwise show Unregistered;
Configure the name of registration server;
Configure SIP register server IP address or Domain Name;
Configure SIP register server signal port;
Configure SIP register account(usually it is the same with the port number
that configured, some special SIP servers will have different port
configurations, then the port configuration needs to be configured to be
numbers, here the configuration account can be arbitrary character string);
Configure password of SIP register account;
Configure the phone number;
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Configure display name;
Configure proxy server IP address or Domain Name(usually SIP will provide
user with service of proxy server and register server which have the same
configuration, so the configuration of proxy server is usually the same with
that of register server, but if the configurations of them are different(such as
different IP addresses or Domain Name), then each server's configuration
should be modified separately);
Configure SIP proxy server signal port;
Configure proxy server account;
Configure proxy server password;
Configure domain realm;
Configure enable/disable register;
Advanced sip setting
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Configure expire time of SIP server register, the default is 60 seconds. If the
expire time that server requires is more or less than that configured by the
phone, the phone can automatically modify it to the recommended time limit
and register;
Configure the NAT keep alive interval;
Configure the User Agent;
Configure the Signal Key;
Configure the Media Key;
Configure the Local Port;
Select the Ring type;
Configure hot-line number of the port. With this number of the port, this
hot-line number will be dialed automatically as soon as off-hook and user
can't dial any other number;
Configure the Conference Number;
Configure the Transfer Expire time
Configure enable/disable Subscribe;
Configure enable/disable Keep Authentication;
Configure enable/disable NAT Keep Alive
Configure enable/disable Via rport
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Configure enable/disable PRACK
Configure enable/disable Long Contact;
Configure enable/disable URI Convert;
Configure enable/disable Dial without register;
Configure enable/disable Ban Anonymous Call;
Call forwarding. The default is Disable; when busy is selected, if the number
dialed is engaged after the phone has received a call, then it will
automatically transfer to the configured number according to the following
configuration; when always is selected, then the phone will directly transfer
all the numbers that dial to this port to the configured numbers;
number configuration of call transfer (CT);
Select the Server type;
DTMF sending mode configuration; three kinds: the above are basic
configurations of SIP.
Enable the phone to use protocol edition. When the phone need to
communicate with phones which is using SIP1.0 such as CISCO5300 and
so on, then it should be configured into RFC2543 to communicate normally.
the default is to enable RFC3261;
Select the Transport Protocol(UDP or TCP);
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Select the RFC Privacy Edition(none,RFC3323 or RFC3325);
Configure Subscribe expire time
Configure enable/disable Conference Number;
Configure enable/disable DNS service;
Configure enable/disable Click To Talk;
Configure enable/disable Signal Encode;
Configure enable/disable RTP Encode;
Configure enable/disable Session Timer;
Configure enable/disable answer with single codec;
Configure enable/disable Auto TCP;
Configure enable/disable Strict Proxy;
Configure enable/disable GRUU;
Configure enable/disable Displayname Quote;
10.3 STUN configuration
SIP STUN is a kind of server that used to realize the SIP's enablement of
NAT, when the STUN server IP of the phone has been configured(generally
the default port is 3478) and Enable SIP Stun has been selected,
conventional SIP server can be used to realize the phone's penetration of
NAT.
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Display the application status of the STUN NAT;
Configure IP address of SIP STUN server;
Configure port of SIP STUN;
Interval time for STUN's detection on NAT type, the unit is second;
Configure Local SIP port;
Select the Sip Line;
Configure enable/disable Use STUN;
10.4 Dial Peer configuration
Bases on this configuration, we can make the phone use different accounts
and run speed calling without swap.
When making deletion or modification, select the number firstly, then click
Delete or Modify , and complete the operation.
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CDX-IPH303-VoIP User Manual V1.3
Display of calling number IP image list;
It is to add outgoing call number, there are two kinds of outgoing call number
setup: One is exactitude matching, after this configuration has been done,
when the number is totally the same with the user's calling number, the
phone will make the call with this number's IP address image or
configuration; Another is prefix matching( be equivalent to PSTN's district
number prefix function),if the previous N bits of this number are the same
with that of the user's calling number(the prefix number length),then the
phone will use this number's IP address image or configuration to make the
call. When configuring the prefix matching, letter "T" should be added behind
the prefix number to be distinguished from the exactitude matching; the
longest length is 30 bits.
Configure destination address, destination is configured 0.0.0.0, it is SIP1
line.
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Configure the protocol signal port, when nothing is input, the default of sip
protocol is 5060;
Configure alias, this is optional configuration item: it is the number to be
used when the other party's number has prefix; when no configuration has
been made, shown as no alias;
Configure the calling mode:IAX2 and SIP;
Configure suffix ,this is optional configuration item: it is the additive dial-out
number behind the number; when no configuration has been made, shown
as no suffix;
Configure the replacing length, replace the number that user input according
to this length; this is optional configuration item.
Of which the alias can be divided into four types, it should be combined with
replacing length to make the setup:
Add: xxx, add xxx before number. in this way it can help user save the
dialing length;
All: xxx, the number is all replaced by xxx; speed dialing can be
implemented, for example, user configure the dialing number as 1, with the
configuration "all" , the actual calling number will be replaced;
Del: delete n bit in the front part of the number, n can be decided by the
replacing length; this configuration can decide the protocol for appointed
number;
Rep: xxx, n bit in the front part of the number will be replaced. n is decided
by the replacing length. For example, user want to dial PSTN(0757-
86228930)by VoIP's Rec/Finish over service, while actually the called
number should be 86757-86228930,then we can configure called number
as 0757T,then rep:86757,and then set the replacing length as 3. So that
when user make a call with 0757 prefix, the number will be replaced as
86757 plus the number and then sent out. It is a convenient thinking mode
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for user to make a call;
Delete selective number IP image;
If user want to modify a certain current number image, first select in the
drop-down menu and then load the image parameter of the said number,
click modify to make modification; of which:
this is the modified number. read-only;
To modify call mode;
To modify destination address; this is optional configuration item;
To modify destination phone port;this is optional configuration item;
To modify alias; this is optional configuration item;
To modify suffix; this is optional configuration item;
To modify replacing length(if rep and del of alias have been configured);
Click submit to go into effect
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11 Phone Configuration
11.1 DSP configuration
CDX-IPH303-VoIP User Manual V1.3
On this page, user can set speech coding, IO volume control, cue tone
standard, caller ID standard and so on.
Configure the first codec;
Configure the second codec;
Configure the third codec;
Configure the fourth codec;
Configure the fifth codec;
Configure handdown time, that is, if the hooking time is shorter than this time,
then the gateway will not consider the user has handdown.
40
Configure input volume;
Configure output volume;
Configure handfree volume;
Configure ring volume;
Configure G729 payload length;
Configure signal standard;
Configure G.722 timestamps;
Configure G.723 bit rate;
Configure default ring type;
Configure enable/disable VAD.
CDX-IPH303-VoIP User Manual V1.3
11.2 Call service configuration
On this page, user can set value added services such as hot-line, call
forwarding, call transfer (CT), call-waiting, three way call, blacklist, out-limit
list and so on.
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Configuration Explanation:
Configure hot-line number of the port. With this number of the port, this
hot-line number will be dialed automatically as soon as off-hook and user
can't dial any other number;
Configure no answer time;
Configure enable/disable Do Not Disturb;
Configure enable/disable Ban outgoing;
Configure enable/disable call transfer (CT); after it is enabled,
automatically.
Configure enable/disable call waiting service; after it is enabled, user can
hold calls of the other party by hooking, with hooking again, and the hold call
can go on;
Configure enable/disable three way call; user can call the other part as the
call origination, after talking, make hooking to hold this part and then
pressSpeed Dial/Conference key to hear the dialing tone, after call
completion to the third party, hooking again to recover the talk with the
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second part, then the three way call concurrently;
Configure enable/disable Accept Any Call;
Configure enable/disable Auto Answer;
Configure add/delete blacklist. If user doesn't want to answer a certain
number, please add this number to the list, and then this number will be
unable to get through the phone.
Configure out-limit list; for example, if user don't want the phone to dial a
certain number, please add the number to this table, and the user will be
unable to get through this number.
11.3 Phone book configuration
On this page, user
can add, delete and modify telephone book.
Some ISP internet may inhibit the phone to register and cancel the register in
succession, so user had better not apply or register and cancel soon in succession
and submit registration repeatedly. Server may stop response of dialogue machine,
then the phone receives no certification of register/cancel login request and
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registration state will show as incorrect!
11.4 Save and Clear Configuration
User can save the current configuration on this page.
The system configuration can be set as factory default configuration on clear
config page and the phone will restart automatically.
12 Security Configuration
12.1 MMI Filter configuration
On the page, user can configure the function of the MMI Filter. This feature
allows only the host within the MMI filter table that they can logon the WEB
page.
On the MMI filter table, user can modify and delete the MMI filter;
Add MMI filter table;
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Configure enable/disable MMI Filter.
12.2 Firewall configuration
On the page, user can configure the function of the firewall.
Select firewall type;
Display firewall input rule table;
Display firewall output rule table; in the table,Src address and Src Mask
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CDX-IPH303-VoIP User Manual V1.3
confirm the range of the source address, For example : Src
addr(192.168.10.10) and Src Mask(255.255.255.255) can confirm that the
host is 192.168.10.10;Dse addr(192.168. 0.120) and Des Mask(0.0.0.0)
can confirm any host; Src addr(192.168.10.20) and Src
Mask(255.255.255.0) confirm any host within the 192.168.10.0 network
segment. So, when the Index is “1” in the output rule table, and the
definition of rule is: the UDP package of the host(192.168.10.10) can’t
was sent to the host(192.168.10.10);when the Index is “1” in the output rule
table, and the definition of rule is:the host(192.168.10.20) can send the
UDP package to any host.
Configure the firewall.
Select the rule of the firewall( input or output);
Select the condition(deny or permit);
Select protocol type(UDP, TCP,ICMP or IP);
Configure port range;
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CDX-IPH303-VoIP User Manual V1.3
Configure source IP address;
Configure destination IP address;
Configure source Mask;
Configure destination Mask;
12.3 NAT and DMZ configuration
On the page, user can configure NAT and DMZ. T function of the NAT is a
network port mapping and the function of the DMZ is a network address
mapping.
NAT configuration
Configure NAT of the Application Layer Gateway(ALG);the protocol includes:
IPSec, FTP and PPTP.
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Display NAT table;
Add and delete NAT table (configure NAT of the Transport Layer, the
protocol includes: TCP and UDP);
Select transfer type;
Configure Inside IP;
Configure Inside port;
Configure Outside port;
Add or delete NAT table;
DMZ configuration
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Display DMZ table;
Configure the DMZ rule.
Configure the outside IP of the DMZ;
Configure the inside IP of the DMZ;
Configure outside IP and inside IP, then click the Add, user can add the DMZ
table.
Select Outside IP;
Select outside ip, and click the Delete, user can delete the DMZ table.
12.4 VPN configuration
On this page, user can save and configure VPN setting.
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Display the VPN IP of the CDX-IPH303;
Select VPN mode, and configure enable/disable VPN;
(1) Select UDP tunnel, and configure VPN:
Select UDP tunnel mode;
Configure VPN server address;
Configure VPN server port;
Configure VPN server group ID;
Configure VPN server area code;
50
Configure enable/disable VPN tunnel;
(2) Select L2TP, and configure VPN:
Select L2TP mode;
Configure VPN server address;
Configure VPN User Name;
Configure VPN Password;
CDX-IPH303-VoIP User Manual V1.3
Configure enable/disable VPN;
13 Upgrade on-line
13.1 Upload WEB page
On this page, user can select the upgrade document (firmware or config file)
on hard disk of the computer directly to run the system upgrade. After the
upgrade has been completed, restart the phone and it will be usable at once.
13.2 FTP/TFTP download
On this page, user can upgrade system and configure files by FTP or TFTP
mode.
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13.3 Configuration Explanation:
Configure upload or download FTP/ TFTP server IP address;
Configure username of the upload or download FTP server. If user select
TFTP mode, username and password are not required to be configured;
Configure upload or download of FTP server password;
Configure upload or download system upgrade document or system layout
file name. It should be noted that system file take .dlf as suffix, configuration
files take .cfg as suffix;
Select server type;
Select Application update type, the phone will upgrade system file;
Select Config file export type and click the Apply, the phone will upload its
configuration files to FTP/TFTP server and save with names of user-defined
configuration files;
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Select Config file import type and click the Apply, the phone will download
configuration files of FTP/TFTP server to the phone and the configuration
will go into effect after restarting;
Output configure file can be edit, delete, or make comment starting by # on
each command. Unit support module upgrade, like if changes made to SIP
configure, others in configure file can be deleted and configuration in unit will
not be affected.
While upgrade unit with modified configure file, please make sure check
each parameter while finished upgrade. In case of anything wrong, please
recover configure under POST mode.
13.4 Configure file encryption
Configure file can be encryption with DOS command:
dsc.exe <key.txt> <e/d> <old configure> <new configure>.
Dsc.exe-encryption software tool
<key.txt>-user made encryption key file
<e/d> e(encrypt), d (decrypt)
< Old configure >former configure file name and path,
< New configure >new configure file name, defined by user.
13.5 Auto-update
Unit can be set as automatically upgrade from desired FTP or TFTP server.
Display the current config version;
Configure unit as follow steps:
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CDX-IPH303-VoIP User Manual V1.3
Input IP add. of desired FTP server.
Input user name of desired FTP server.
Input user password of desired FTP server.
Input name configuration file. Software version must be different for each
upgrade file.
If configuration file is encrypted, password needed.
Chose server type as either FTP or TFTP.
Set auto-upgrade interval duration.
Chose auto-upgrade type.
13.6 Configuration files WEB download
On this page, user can directly select the configuration files on the hard disk
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of the computer, and then make modification to the system configuration,
after the download, restart the phone and the configuration will go into effect.
14 System management
14.1 Account management
On this page, user can set the keyboard password, the same time, user can
add and delete users according to own needs and can modify user's
authorities there have been.
14.2 Configuration Explanation:
display of phone user account list;
To add phone account; it will be shown at lower part of page as the following
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CDX-IPH303-VoIP User Manual V1.3
figure, of which:
Add new accounts;
account level; root possesses
authorities to modify configuration,general possesses read-only authority;as
corresponding password of the additive account;
As second confirmation of password, to ensure correct setup of password;
Clicks submit to go into effect.
Select users that you want to delete in the drop-down menu, click Delete.
To modify the chosen accounts, need to select account first, click Modify,it
will be shown at lower part of page as the following figure, of which:
The modified username;
Modify user authorities;
Modify user password;
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CDX-IPH303-VoIP User Manual V1.3
Make confirmation of the modified user password;
Submit the modification;
Owing to the phone's default account: accounts of the administrator
level-admin and the ordinary level-guest are all weak account and weak
password, the username and password will be easily to be guessed on
public network, so the user had better modify the administrator and ordinary
user.
Enter with manager level when making modification, create a administrator
account and a browse account (you'd better not set the name as admin,
administrator, guest, etc.),set password and then save configuration,
entering with new manager account, delete default manager and browse
account and save configuration, security will be enhanced!
14.3 Time zone configure
On this page, user can save and configure time zone setting.
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CDX-IPH303-VoIP User Manual V1.3
Configure the desired time zone.
15 Configuration via Telnet
In DOS window, input telnet 192.168.10.23, enter:
Then input USER NAME:admin PASWORD:admin
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15.1 Basic Command
input “help” or “ ?”under terminal to check all sub-terminal and globe
command; input “help” or “?”under command to check parameters;
input “!” or“exit”to quit former path. 。
while partly input terminal or command, press“TAB”, system will auto finish
balance command or list all option.
each help is with comment as <command> or <node> to identify.
parameter of each command including two types:“required” and“optional”:
all “required”start as “-”; all option,start as“_”. Also partly input available like
point 3.
after configure, make sure save with command “write”, other wise, all setting
lost while re-start.
15.2 Command structure
Root terminal with structure as
#
---config
---debug
---download
---language
---password
---setdefault
---show
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---telnet
---trancert
---update
---upload
Most command parameters are under terminal “config”, which structures
as follows:
<config>#
---accesslist
---dialpeer
---digitalmap
---fastethernet
---mmifilter
---nat
---port
--qos
---syslog
---time
---user
---voip
---vpn
15.3 Structure of Configuration terminal
Access list firewall configuration
Path:<config-accesslist>#
Port configuration
While input PORT under terminal config, the configuration will valid to all
ports, if input as PORT X, valid only to port X. Some parameter only valid to
some port, then PORT X is needed, otherwise, error report as “Error:
Missing parameter”.
Path: <config–port>#
[no] accept-relay - <command>Set accept relay mode
[no] calltransfer - <command>Enable/Disable call transfer
[no] callwaiting - <command>Enable/Disable call waiting
codec - <command>Set Codec
[no] fastcalling - <command>Set fastcalled number
handdown - <command>Hand down delay
[no] in-limit - <command>Set the number which will be not
accepted
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[no] input - <command>Set Input gain
[no] out-limit - <command>Set the number which can not be
dialed
[no] output - <command>Set Output gain
[no] ringvolume - <command> set ring volume
show - <command>Show port configuration
[no] shutdown - <command>Disable/Enable the port
signalmode - <command>Set signal mode
[no] threetalk - <command>Enable/Disable threetalk
QOS
Path: <config-qos>#
[no] 8021p - <command> set 802.1P Priority
[no] diffsevenable - <command> Enable/Disable DiffServ
[no] diffsevvalue - <command> set DiffServ Value
show - <command>Show QOS configuration
[no] vlanid - <command> Set VLAN ID
USER management
Path: <config–user>#
[no] entry - <command>add a user with given level
password - <command>Change password
show - Show all users
Other commands
Setting module debug message level ---debug all xxx
Setting MGR module debug message level ---debug MGR xxx
Setting SIP module debug message level ---debug sip xxx
Setting IAX2 module debug message level ---debug IAX2 xxx
Remove module debug message level ---debug no all
Remove MGR module debug message level ---debug no MGR
Remove SIP module debug message level ---debug no sip
Remove IAX2 module debug message level ---debug no IAX2
Reset to default ---setdefault
Reset all to default ---setdefault all
Check message of some module ---show xxx
Update present password ---password
Telnet remote login ---telnet x.x.x.x
Use special Telent port---telnet x.x.x.x –port xxx
Telnet quit ---logout
save ---write
re-start ---reload
get help ---help
quit ---exit
clear screen displa ---clear
PING host ---ping x.x.x.x
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CDX-IPH303-VoIP User Manual V1.3
trace ---trancert x.x.x.x
If user has accounts of a certain SIP server and each account has different
password, then user should add each account and its corresponding
password to the account& password table.
Configure display of account & password list;
Click Add to add account and password, it is shown as the following figure:
Configureadditiveaccounts
Configure additive passwords
Click submit to submit the configuration, click return to cancel the
configuration and return;
Select accounts that you want to delete from the drop-down menu, click
delete.
Select drop-down menu to select accounts that want to modify, click load to
load the configuration and then click modify to modify;
Accounts to be
Modified, read-only;
Passwords to be
modified;
Click submit to
submit, click return to cancel the modification and then return.
1095 Budapest, Mester utca 34.
Tel.: *218-5542, 215-9771, 215-7550,
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