Telematrix IP550 User Manual

IP550 VoIP Phone
User Manual
TeleMatrix, Inc
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1.1 Overview of Hardware......................................................................................5
1.2 Overview of Software
2 Keypad of IP550
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2.1 Function Table of Keyboard
2.2 Keyboard function and designed catalog
3 Set the Phone Through WebBrowser
3.1 Login
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3.2 Current state....................................................................................................13
3.3 Network............................................................................................................13
3.3.1
Wan Config
Lan Config.................................................................................................13
3.3.2
3.4 VoIP
3.4.1
3.4.2
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SIP Config
Iax2 Config................................................................................................18
3.5 Advance
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DHCP Server............................................................................................19
3.5.1
NAT...........................................................................................................20
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
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STUN
Net Service...............................................................................................22
Firewall settings
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VLAN Configuration
Digital Map................................................................................................24
Call Service Settings
Memory Key..............................................................................................27
MMI Filter................................................................................................27
Audio Settings
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VPN.........................................................................................................28
3.6 Dial-Peer dial rule setting
3.7 Config Manage
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3.8 Update Firmware ............................................................................................31
3.8.1
Update
Auto Update..............................................................................................32
3.8.2
3.9 System Manage
3.9.1
Account Manage
Syslog Config............................................................................................33
3.9.2
3.9.3
Phone Book
Time Set....................................................................................................34
3.9.4
MMI SET...................................................................................................35
3.9.5
3.9.6
Logout & Reboot
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4 Operating Method for Dialing
4.1 How to dial IP Phone
4.2 Set the Phone to Server
Set WAN Interface....................................................................................36
4.2.1
4.2.2
SIP Setting
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IAX Setting................................................................................................39
4.2.3
4.3 How to Use Dialing Rules...............................................................................39
4.4 Voice mail
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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 “SYS/IP” 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 conditio n 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.
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Function
1.
Support two SIP server working at the same time
2.
Provide a Backup SIP Server
3.
Support NAT, Firewall
4. Support DHCP assign IP address, etc automatically
5. Support PPPoEused while connecting ADSLcable modem
6.
It can update the program through HTTP ,FTP and TFTP
7.
Check the dynamic voice; Soft the noise; Buffer technique of voice
8.
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
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1. Introduction
This is the user manual of IP550. Some configuration should be done before use the IP550 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
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Call Forward 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 IP550
2.1 Function table of keystroke
Name Status Function/Display
Up config Choose the Config PageRoll up
Down config
Choose the Config Page(Roll down)
SYS/IP config System information
Enter config Confirm/get into the submenu
Exit Exit from the current status
Menu
Del
On-hook
Call
Get into the setting menu
Config Modify the current configuration
Dialing Delete the frontal digit
Call log On-hook Call record
FWD Call Call forward Redial Phone
Call
Dialing
Call the number dialed last time
On-hook Get into the phonebook book Conf Call Three-way communication
Volume + Call Increase the volume
Volume - Call Decrease the volume
Hold Call Hold on DND On-hook Do not disturb Mute Call Mute
Headset
On/Off
Switch for the handset
Speaker Speaker
M1~M11
Voicemail
On-hook
Dialing
On-hook
Dialing
11 speed dial numbers
Pick up voicemail
Firmware Upgrade
TFTP
HTTP
FTP
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1
2
3
4
5
6
7
8
9
0
*
Dialing “1”
Config “1”, “space”, “@”, “_”, “-”, “/”, “%”
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”, “*”, “#”, “$”, “&”, “?”, “!”, “<”, “>”
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 System information and IP address query Every time you press “SYS/IP”, there will be one basic System information showed on LCD.
1.Numberdisplay the local phone numbers
2. Static/DHCP/PPPoE:display IP address of WAN port
3. SIPdisplay whether to register SIP account or not
4. SIP SERVERdisplay SIP server address
5. Gateway:display subnet mask
6. NetMaskdisplay gateway
2.1.5
Three-way conference call
Suppose the user of the IP550 phone is A, and if user B through VoIP
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phone to call user A, user B need to m ake 3-way co nference call with user A and C, then user A ca n p ress “Conf” key to hold the conversation with user B, then call user C, during the conversatio n with C to pre ss 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 T ransfer The “Call Transfer”in the “Call ServerEnable”should be in chosen mode(the
default one is chosen mode). Suppose the user of IP550 is user A, and user B called us er Aduring the communic ation, 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.
2.1.7 Do not disturb function 1 Enable the Do Not Disturb functio n:
Press "DND" key under the On-Hook status, the indica torlight for "Do not disturb" will be lighted, and then the phone will enable the function of Do Not Disturb. 2、 Disable the Do Not Disturb function:
Press "DND" key under the On-Hook status,the indicatorlight for "Do not disturb" will be turned off, and then the phone will disabl e the function of Do Not Disturb.
2.1.8 Call Record Press "Call log" key to enter Call Reco rd menu und er On-Hook st atus, then press "up/down" for entering into submenu. 1、 Dialed
Check the Outgoing Call record.
2、 Received
Check the Incoming Call Record.
3 Missed
Check the Missed Call record. If the phone has missed calls, then one can press "Enter" key directly for checking the missed calls under the status of On-Hook.
2.1.9 Phone book
Press "Phone book" key to enter into Phone book menu under ON-Hook status, and then press "up/down" key for entering into the submenu: 1、current
The current phone book. Pressing Enter key for entering, then Press up/down key to check the current contact and information.
2、Add
Adding contact. Press Enter key to enter into the menu, then input and confirm the new contact name ,number, and ring type according to the
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prompt, and you will add the new contacts into the menu.
3. Search Searching for contact Press key “ENTER” to enter, and then input the name of contact and be confirmed, that is contact searched and contact information displayed.
2.1.10 The use of Memory key
1、Method for Storing a) Press "memory" key without any storing number, then press "Del"key to input the number you want to stored, then press "Enter" key directly to store the number after confirming the number carefully. b) Press "Menu" key and input the password, then enter into systemsave, and store the setting afte r confirmation.
2、Method for modification
a) After entering into the setting menuOther setting Memory, find the keystoke option that need to be deleted/modified, and press "Del" to delete, then input the modified number, and press "Enter"key to store.
b) Back to the main menu, then enter into systemsave, and store the setting after confirmation. 3Method for deleting a) After entering into the setting menu  Other Setting Memory,
find the keystoke option that need to be del eted/modified, and press "Ente r"to store after press "Del", then the keystoke option can be deleted.
b) Back to the main menu, then enter into systemsave, and store
the setting after confirmation.
2.1.11 The earphone switch: After pressing the "Handset On/Off" key, if the indicator light lights, the voice will be forwarded to the earphone by the phone, and the voice of Handset or Speaker will be cut out at the same time. If the indicator does not light, the voice will be forwarded to the Handset or the Speaker, and the voice of earphone will be cut out at the same time.
2.2
Functions and setting catalog
1Press down the "Menu" key to enter the setting state under the ON-Hook status, and the default password is 123, then press "Enter" for entering. Use "UP/Down" key to choose the catalog, then press "Enter" to enter into the catalog, and "Exit" to exite.
2、 When modifying the setting, press “Del” to enter modi fication s tatus, and “0” is to make no choice, “1” is to make choice, “Enter” is to confirm the modification, “Exit” is to quit the modification. After finished the modification setting, we will save it on “Save” menu. After rebooting all the settings will be go into effect.
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3Menu catalog
1) Network
2) Call Feature
3) SIP
4) DSP
5) System
6) Other Setting
2.2.1 Network:
2.2.1.1 LAN
1) Bridge Mode
2) IP
3) Netmask
4) DHCP Server Switch
  DNS Relay
5) NAT
Switch FTPalg
PPPTPalg
2.2.1.2 WAN
1) Status
2) Static Net
IP
NetMask
  Gateway
DNS
DNS2
3) PPPoE User name
  Password
4) QoS
2.2.2 Call Feature
2.2.2.1 Phone-number
1) Public SIP
2) Private SIP
2.2.2.2 Limit-List
1) Current
2) ADD
3) DEL
2.2.2.3 Black-List
1) Current
2) ADD
3) DEL
2.2.2.4 FastCall
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2.2.2.5 Three Talk
2.2.2.6 Call-Transfer
2.2.2.7 Call-Waiting
2.2.2.8 Call-Forward
1) Condition
2) SIP Transfer Num
Transfer IP
  Port
2.2.2.9 Dial-Rule
1) End With #
2) Fixed Length Switch
Length
2.2.3 SIP
2.2.3.1 Reg Status
1) Public Reg
2) Private Reg
2.2.3.2 Reg Switch
1) Public
2) Private
2.2.3.3 Server
1) Public
2) Private
2.2.3.4 Domain
1) Public
2) Private
2.2.3.5 User Agent
1) Public
2) Private
2.2.3.6 Detect-server
2.2.3.7 Dtmf-mode
2.2.3.8 Interval-time
2.2.3.9 Swap-server
2.2.3.10 RFC-version
2.2.3.11 Signal-Port
2.2.3.12 Stun
1) Switch
2) Addr
3) Port
4) Expire Time
2.2.4 DSP
2.2.4.1 Codec
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2.2.4.2 Handdown-time
2.2.4.3 Dtfm-Volume
2.2.4.4 Input-Volume
2.2.4.5 Output-Volume
2.2.5 System
2.2.5.1 Save
2.2.5.2 Reboot
2.2.5.3 Set Default
2.2.6 Other Setting
2.2.6.1 Syslog
1) Switch
2) Server-IP
3) Server-Port
2.2.6.2 Menory
1) Memory-1
2) Memory-2
3) Memory-3
4) Memory-4
5) Memory-5
6) Memory-6
7) Memory-7
8) Memory-8
9) Memory-9
10) Memory-10
11) Memory-11
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 IP550.
3.1 Login:
The default user name and password are admin/admin and guest/guest.
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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 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.
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Configure Static IP:
----Enable Static;
----Set IP550’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, IP550 will automatically obtain WAN
port network information from your DHCP server.
Configure PPPoE:
----Enable PPPoE
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----PPPoE server: Enter “ANY” if no specified from your ITSP.
----Enter PPPoE username and pin in the username and password.
IP550 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.
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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:
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Register Server Addr: Register address of public SIP server
Register Server Port:
Register port of public SIP serverdefault port is 5060
Register Username: Username of your SIP account (Always the same as the phone number)
Register Password: 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: Proxy Username:
Signal port of SIP proxy
proxy server username
Proxy Password: proxy server password Domain Realm:
SIP domain, enter the sip domain if any, otherwise IP550
will use the proxy server address as sip domain.
Local SIP port: Phone Number:
Local SIP register port, default 5060
Phone number of your SIP account
Enable Register: Enable/Disable SIP register.IP550 won’t send register info
to SIP server if disable register.
Enable Message Waiting:The configuration allows/forbids Message Waiting.
Advanced SIP Setting
Register Expire Time: register expire time, default is 60 seconds. IP550 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. AlwaysThe caller always forward to the appointed phone.
for detail.
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Forward Photo Number Detect Interval Time: Co-work with the Auto Detect Server, if Auto Detect Server is
enable, IP550 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 TypeThe 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:
to communicate to devices (such as CISCO5300) using the SIP 1.0. Default is RFC 3261.
call the forwarded phone number.
Current IP550 SIP version. Set to RFC 2543 if the gate need
3.4.2 Iax2 Config
Setting page of public IAX server:
IAX Server Addr:
Register address of public IAX server
IAX Server Port: Account Name:
number)
Register port of public IAX serverdefault port is 4569 Username of your SIP account (Always the same as the phone
Account Password: Password of your IAX account. Local port: Phone Number:
Voice mail number:
mail is letters which you can not input with the ATA, then you use the number to sta nd for your username Voice mail text: if IAX support voice mail, config the domain name of your mail box here.
Echo test number:
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 G.729: Using G.729 speech coding mandatory consultations IAX2(Default Protocol):
choose SIP as default
Signal port of local, default port is 4569
Phone number of your IAX account
If the IAX support voice mail, but your username of the voice
If the platform support e cho te st, a nd t he number i s te st for m , th e
echo test number in text format
IAX refresh time
enable or disable register
Set IAX 2 as the default protocol , if not the system will
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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 Tabledisplay the IPMAC corresponding table that the server
distributed.
Lease Table Name: Start IP: End IP: End IP of lease table. Network device connecting to the IP550 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.
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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.
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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 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
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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.
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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.
IP550 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.
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3.5.6 VLAN Configuration
IP550 phone implement QoS based on 802.1p, The QoS is used to mark the network
communication priority in the data link/MAC sub-layer. IP550 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 IP550 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.
802IP 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. IP550 support below digital map:
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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. IP550 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.
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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. IP550 immediately dials this number af ter hook-of f
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, IP550 refuse the incoming call when the
called number is different from IP550’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
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3.5.9 Memory Key
This page layout shows the number setting of Voice mail and speed-dial key.
3.5.10
When MMI filter is enable. Only IP address within the start IP and end IP can access IP550 phone.
MMI Filter
MMI filter is used to make access limit to IP550 phone.
3.5.11 Audio Settings
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CODEC: select the prefer CODEC; support ulaw, alaw, G729 and G7231 5.3/6.3 Signal Standard: Handdown Time:
Signal standard for different area.
hand down detect time.
Input Volume: Handset in volume. Output Volume:
Handset out volume.
Handfree Volume: Hand free volume G729 Payload Length: VAD:
Enable/disable Voice Activity Detection
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
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 the group ID of UDP VPN
Server Area Code
After VPN registered successf ully, VPN server will give an IP aggress to the
register to the address of VPN server
Register to the port of VPN server
the area code of VPN server L2TP VPN Server Addr: register to the address of VPN server
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VPN User Name
L2TP VPN username
VPN Password UDPTunnel: use the UDP to visit VPN
L2TP: 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.
use the L2TP to visit VPN
L2TP VPN password
Dial-Peer dial rule setting
3.6
Please refer to How to use dial rule
for detail.
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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.
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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, IP550 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 IP550 phone support FTP and TFTP auto update, the gateway will auto obtain the
configure file from your update server if configured. To obtain the original configure 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.
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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 updateUpdate after reboot:
auto update after rebootUpdate 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>>Version:1.0000
<GLOBLE CONFIG MODULE> ConfFile Version:6
User may edit the configure file version to:
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<<VOIP CONFIG FILE>>Version: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 IP550.
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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.
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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
Logoutlogout 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 reb ooted, you can enter into this l ayout and cl ick
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“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 bein g made a proper set ting on your p hone. Ple ase 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 NetworkWAN Config layout phone of the net
port:
IP550 could be connected to Internet by using the static IP, DHCP IP, or PPPoE
dialing.
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Configure Static IP:
----choose static;
----fill in the IP address of IP550 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 respectively.
Use the configure to dynamic obtain IP to get IP address:
----choose DHCP option. Now, if the network has DHCP server, then IP550 will get IP address, Netmask, Gateway, Primary DNS and Alter DNS from this DHCP server automatically.
Use PPPoE dialing to connect the Internet:
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----choose PPPoE option.
----please fill in the account and password which PPPoE have dialed in the PPPoE Username and Password. So IP550 could connect the Internet through 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:
Register Server Addr: Register Server Port: Register Username:
phone number)
VoIP  SIP Config
to set the layout config and sip account
Register address of public SIP server
Register port of public SIP server
Username of your SIP account (Always the same as the
default port is 5060
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Register Password: Phone Number: Phone number of your SIP account
----choose Enable Register You can dial V oIP pho ne 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 V oIP pho ne 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
IP550 provide flexible dial rule, with different dial-rule configure, user can easily
Register address of public IAX server Register port of public IAX serverdefault 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
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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 new dial rule. Below is the detail setting of the dial-rule:
Phone Number:
The Number suit for this dial rule, can be set as full match or prefix match. Full match means that if the number user dialed is completely the same as this number, the call will use 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:xxxreplace 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:
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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, su it for LAN application
4.4 Voice mail
When there is a mail, voice mail LED would be flickering, and LCD would display “New message”. After listening the message, voice mail LED would stop flickering, and “New message” would disappear from LCD.
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