5520 User Manual V1.
TABLE OF CONTENTS 1st 5520’s Network Features.........................................................5 1 2 3 4 5 6 7 8 9 10 11 The View....................................................................................................5 Interfaces...................................................................................................5 Hardware................................................................................................... 5 Software............................................
13 14 15 16 17 vport...................................................................... ..................................26 Click to dial...............................................................................................26 SMS function........................................................................................... 27 Default Password.....................................................................................28 Check the Phone’s IP.........................................
8.2. Phone Book...............................................................................................60 8.3. Multi Line Set.............................................................................................61 8.4. Function Key Set.......................................................................................62 8.5. Syslog Config.............................................................................................63 8.6. Time Set..............................................
、 、 5520 、 ¾ ¾ ¾ ¾ ¾ ¾ Power: Output Power: 12VDC, 500mA DC WAN: RJ45 port LAN: RJ45 port Extended interface for BLF module: 2 Headset jack : RJ9 port Handset jack : RJ9 port 、 ¾ LCD: 128×64 dot matrix ¾ FLASH: 4M ¾ RAM: 16×16M ¾ LED indicator ： 1 Status Light , 9 BLF indicator, 1 voicemail indicator, 1 headset indicator, 1 mute indicator, 1 handfree indicator 5
、 ¾ ¾ ¾ ¾ ¾ ¾ ¾ ¾ ¾ ¾ ¾ ¾ ¾ ¾ ¾ ¾ ¾ ¾ ¾ ¾ ¾ ¾ ¾ ¾ Sip 2.0 (RFC3261) and other related SIP RFC 4 lines SIP 1 line IAX2 STUN Jitter Buffer(200ms),VAD,CNG G.711A/u, G722, G.723, G.726-32, G.729 Codec G.168 compliant 96ms echo cancellation Support SIP domain，SIP authentication (none，basic,MD5),NDS.
¾ ¾ Qos support Diffserv Support Network command tool: include ping, trace route, telnet 、 ¾ Support safe mode(POST Mode) and firmware updating under safe mode ¾ Support different level user management ¾ Configuration via web , keyboard and command ¾ Support multi language (LCD support Latin language system, web support all languages) and easy dynamic switch between different languages ¾ Firmware and configuration updating via HTTP , FTP and TFTP ¾ Support system log and call log ¾ Firmware and configurat
The letter “e” is the first letter of “environment: and “electronic”. The rim is a round with two arrow, stands for recycle. The number 20 stands for the years of environment protection. Please note the years of environment protection is not discarding year nor usage life.
、 1、Desktop position: A、Put the bottom side of the IP phone upside and press the plate with letter “PUSH” into the slot, please refer the picture as below: B、Press the other plate into the slot in accordance with the direction of the arrow 9
C、Repeat A and B. It is the right picture of putting on desk after fixing the two feet below: D、Disassemble the feet: Press the plate with word “PUSH” and pull the feet with the direction of arrow.
C、Repeat A and B. It is the picture of wall mounting after fixing the two feet below: D、Disassemble the feet way: Press the plate with word “PUSH” and pull the feet with the direction of arrow.
Describe of the buttons and Screen: Soft buttons Press to select an feature shown in the soft button features Soft button features Shows available choices based on current phone function displayed on the last line of LCD screen Status Shows the phone status, if the phone is standby, the LED is with light. If there is income calling, the LED will flicker. ¾ If the phone is starting ,the LED is flicker ¾ If the phone is standby, the LED is off ¾ If there is income calling, the LED will flicker.
Volume buttons Adjust the volume Speakerphone Pick up and hung up on the speakerphone mode, when pick button up by speakerphone, the LED of the button is on. Mute button Mute the handset, headset or speakerphone by press the Mute button; this prevents the person on the active call form hearing what you or someone else in the room is saying. To cancel the Mute function, press the Mute button again. If Mute the voice, the LED is light on this button. Headset button Pick up and hung up on headset mode.
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、 、 When there is an incoming call, 5520 will remind user with ringing. There are 5 ways to answer the call A、Answer by handset Pick up the handset and talk with the caller. If you want to hang up, just put back the handset. B、Hand-free mode Press the hand-free button in the phone and talk with callers by built-in Microphone and Speaker. If you want to hang up, please press the hand-free button again.
C、Answer by earphone Keep your earphone connected with the RJ9 earphone jack, when there is an incoming call, press the earphone button on the IP phone and talk with the caller. If you want to hang up, please press the earphone button again. D、Handset to hand-free When you are phoning with the handset and want to phone with hand-free mode, please press the hand-free button and put down the handset.
contact person. e、When you find the certain contact person, press" OK" to show the details. f、Press "Edit" to edit the number or press" Dial" to call. 、 It’s method for the phone in standby mode to dial number immediacy. The method is as below： A、Dial-up the number in standby mode B、Push soft button “dail”, ”#”key or hang up directly to send the dial number. C、Push soft button to save the number in telephone directory. 、 5520 IP phone supports 4 Sip lines.
3) Earphone Hang up When use Earphone calling, Press the soft button “headset” to hang up. 4) Hang up one line call When 2 lines call simultaneous, press soft button “SWIT” to choose the line which you want to hang up, then press soft button “#” to end the call. In the mean time, it will automatic switch to another line and continue call. Moreover, user can redial-up or accept the second call. Hang up with “#” is invalidation when only one line call. 、 ¾ Blind Transfer User A.B.
User A.B.C, assume B is 5520 Ip phone: 1) 2) 3) 4) 5) When A Calls B with B receives. B presses soft button “Xfer” when A is calling. B dials C’s number. After dialing C, B directly Presses soft button “Xfer”, then transfers the call to C. C receives the phone, starts to talk to A. Remarks: To carry out this function, IP Phone must work with Call waiting and call transferfunction; meanwhile Sip server must support RFC3515. 、 User can hold the current call by pressing soft button “Hold”.
2) Press navigation button to browse missed call history. 3) Choose the missed call record, press “OK” soft button to browse the specific information of the record. 4) Press “Dial “soft button to call back it. 5) Press “Edit“soft button to edit the item and save number. 6) Press “EDia “soft button to revise the records and press soft button “dial” to call this number. ¾ Incoming call Method 1, 1) Press the "MENU" button. 2) Press the navigation button to choose “call history” and then press OK button.
、 Call pickup is simulated from “Pickup” function processes from IPPBX. When A call B with no reply after ring tones, C could pick up the call from A for B by inputting the prefix and B’s phone No. C needed to set the dial peer with prefix code as follow To refer *1* as the set prefix code, C could get the call from A to B by dialing *1*+B, *1* prefix could be freely set as long as no confliction with other dialing rules. Del Length is the digits of the prefix.
*3* is the prefix. Then A could make the redial function via dialing *3* + B's number. *4* is the prefix. Then A could make the unredial function via dialing *4* + B's number. User could set any prefix if it is compliant with present dial rule. Del Length is the digits of the prefix. 、 Vport makes more flexible calling application. Eg. It could forward a call from Line 1 to one account of Line 2 after configuring forward type and number@line via web interface.
、 ¾ Create new SMS 1) Press MORE (soft button 4) 2) Press SMS(soft button 2） 3) Press New(soft button 1) 4) Edit SMS context and you can switch the input method by press # such as ABC(capital letters) ，abc ( lower case letters) , 123 ( number). 5) When the edit is done, press Send(soft button 2）and input receiver’s phone number.
、 There are 2 models to set the authority of web accessing and command line: Guest model and Admin model. User could view and configure all items in Admin mode. While guest couldn’t change the SIP (1-2) and IAX2 configuration as well as server address and port but only access and view the information.
、 This page shows the IP phone working status. The network part shows the connection status of WAN and LAN. Phone Number part shows the phone number and register status for Line1、Line2 and IAX2. The Version shows the current firmware version. 、 There are 3 ways to connect to the internet DHCP, Static and PPPoE, please choose one according to your own situation.
： ¾ Active IP： IP phone’s address. ¾ Current Net mask：network net mask. ¾ MAC Address：MAC of IP phone. ¾ Current Gateway：the IP address of the router. B、If your ISP provide you with the fixed IP address, please choose static and fill in the correct information of IP Address、Net mask、Gateway、Primary DNS etc. If you do not know it please refer to your ISP provider or network management stuff. The reference picture is as below.
： ¾ ¾ ¾ ¾ ¾ ¾ Static IP Address: fixed IP address. Net mask: LAN net mask. Gateway: Gateway IP address. DNS Domain: input DNS domain name if it’s provided. Primary DNS: Primary DNS address. Alter DNS: Alternative DNS address. C、when you use PPPoE to get IP address，please select “PPPoE”，and input ADSL account information as below picture： Parameters： PPPoE Server: sever name, if the ITSP have no special requirements, please keep"ANY" as default. Username: ADSL account username.
： ¾ LAN IP：config LAN static IP. ¾ Net mask: LAN net mask. ¾ DHCP Service: enable LAN DHCP Server, need to reboot to make it available. ¾ NAT: Network Address Translation. ¾ Bridge Mode: Select Bridge Mode or not: If you select Bridge Mode, the phone will no longer set IP address for LAN physical port，LAN and WAN will join in the same network. Click “Apply”, the phone will reboot.
¾ Register Status: SIP server registration status, if succeed display Registered， or else display Unregistered. ¾ Server name: SIP server name, if no special requirements just keep it as blank. ¾ Server Address: SIP server address, support both IP address and domain name. ¾ Server Port: SIP server port, default is 5060. ¾ Account Name：SIP account name. ¾ Phone Number：SIP account phone number, if leave it as blank, no registration information will be sent out.
¾ Proxy Username: Input your SIP register account name. ¾ Proxy Password: Input your SIP register password. ¾ Domain Realm: config SIP local domain. If the server does not have special requirements for the local domain of SIP terminal, the local domain can be the same as SIP server domain. The user can also leave it as blank; the system will take SIP server domain as the domain realm. ¾ Enable Register: Enable or disable registration. ¾ Register Expire Time: register expire time, default is 60 seconds.
¾ Auto Detect Server Interval: Set examining interval of the server, default is 60 seconds. ¾ User Agent: Set the user agent if have, the default is VoIP Phone 1.0. ¾ Signal Key: Signal encryption Key. ¾ Media Key: voice stream encryption Key. ¾ Local Port: Local SIP signal port，default as 5060. ¾ Hotline Number: Set hot line number of each line. ¾ MWI Number: Set SIP1 voicemail Number.
¾ Busy：If the phone is busy, incoming calls will be forwarded to the appointed phone. ¾ No answer： If there is no answer, incoming calls will be forwarded to the appointed phone. ¾ Always: Incoming calls will be forwarded to the appoint phone directly. The phone will prompt the incoming while doing forward. ¾ Forward Phone Number: Appoint your forward phone number. ¾ Server Type: Select the special type of server which is encrypted, or has some unique requirements or call flows.
¾ Enable Subscribe: Enable Subscribe: Overtime of resending subscribe packet. Suggest using the default config. Above is the IAX server configuration page ¾ IAX Server Addr: Register address of public IAX server. ¾ IAX Server Port: Register port of public IAX server，default port is 4569. ¾ Account Name: Username of your SIP account (Always the same as the phone number). ¾ Account Password: Password of your IAX account. ¾ Local port: Signal port of local, default port is 4569.
¾ Voice mail text: if IAX support voice mail, config the domain name of your mail box here. ¾ Echo test number: If the platform support echo test , and the number is test form , the config the test number to replace the text format The echo test is to test the error status of terminals and platform. ¾ Echo test text: echo test number in text format. ¾ Refresh time: IAX refresh time. ¾ Enable Register: enable or disable register. ¾ Enable G.729: Using G.729 speech coding mandatory consultations.
¾ Lease Table Name: Lease table name. ¾ Lease Time: DHCP server lease time. ¾ Start IP: Start IP of lease table. ¾ End IP: End IP of lease table. Network device connecting to the 5520 LAN port can dynamic obtain the IP in the range between start IP and end IP. ¾ Net mask: Net mask of lease table. ¾ Gateway: Default gateway of lease table. ¾ DNS: default DNS server of lease table. o Press “add” to apply, will added DHCP lease table.
DMZ config： In order to make some intranet equipments support better service for extranet, and make internal network security more effectively, these equipments open to extranet need be separated from the other equipments not open to extranet by the corresponding isolation method according to different demands.
The setting page as below: ¾ IPSec ALG: It is an encryption technology. Select it to enable IPSec ALG, the default is enable. ¾ FTP ALG: FTP is a service of connection layer which can transform intranet IP into extranet IP when intranet IP is sending out packet. Select it to enable FTP ALG, the default is enabling. ¾ PPTP ALG: Select it enable PPTP ALG, the default is enable. Shows the NAT TCP mapping table Shows the NAT UDP mapping table.
¾ Transfer Type：Select the NAT mapping protocol style, TCP or UDP. ¾ Inside IP: Set the IP address of device which is connected to LAN interface to do NAT mapping. ¾ Inside Port: Set the LAN port of the NAT mapping. o Outside Port: Set the WAN port of the NAT mapping. After finish setting, click the Add button to add new mapping table. Click the Delete button to delete the selected mapping table. ¾ DMZ Table: Shows the outside WAN port IP address and the inside LAN port IP address.
¾ HTTP Port: set web browser port, the default is 80 port，if you want to enhance system safety，you'd better change it into non-80 standard port； Example: The IP address is 192.168.10.88. and the port value is 6090, the accessing address is http://192.168.10.88:6090 ¾ Telnet Port:Set Telnet Port, the default is 23. You can change the value into others. Example: The IP address is 192.168.1.88. the telnet port value is 6023, the accessing address is telnet 192.168.1.
¾ in_access enable: Select it to Enable in_ access rule. ¾ out_access enable: Select it to Enable out_ access rule. o Firewall Input Rule Table: Firewall input rule, as the picture config is deny o 192.168.1.2 ping 192.168.10.2, but ping 192.168.10.0/24 beside o 192.168.10.3 is ok. ¾ Firewall Output Rule Table: Firewall output rule, as the picture config is the phone ping 192.168.1.70 was deny. ¾ Input/output: Specify current adding rule by selecting input rule or output rule.
¾ VLAN Enable：Before select it to enable VLAN, you need enable Bridge mode in LAN config. ¾ VLAN ID Check Enable：Enable VLAN ID check by selecting it. After enable VLAN ID check, if VLAN ID of a data package is not the same with the phone’s or a data package do not have VLAN ID, the data package will be discarded. ¾ Voice/Data VLAN differentiated：After enable VLAN, system will set packets with different type of VLAN ID.
disable the DiffServ, then system will not distinguish the voice and data, all packets will use the Voice VLAN ID as the tag. 3) Enable VLAN, if set Voice and Data VLAN differentiated as tag differentiated and enable the DiffServ, then system will distinguish the voice and data and add the VLAN ID each other.
Digit map is a set of rules to determine when the user has finished dialing. 5520 support below digital map: ¾ End With “#”: Use # as the end of dialing. ¾ Fixed Length: The call will be sent out automatically when the length of the number you dial reaches the fixed one. For example if you set number of 11 here, when you dial 11 digits the call will be sent out immediately. ¾ Timeout: Specify the timeout of the last dial digit. The call will be sent after timeout.
¾ STUN NAT Transverse：STUN NAT Transverse status true or false. ¾ STUN Server Addr: configure stun server address. ¾ STUN Server Port: configure stun server port default 3478. ¾ STUN Effect Time: stun detect NAT type interval time .If NAT found a link inactive for a certain time , it will close the link so you need to send a packet within a interval tome to keep the link alive. ¾ Local SIP Port: config local SIP port, default as 5060. ¾ Use Stun：enable/disable SIP STUN.
¾ Enable Call Transfer: Enable Call Transfer by selecting it. ¾ Enable Call Waiting: Enable Call Waiting by selecting it. ¾ Enable Three Way Call: 3 way conference call. ¾ Accept Any Call: If select it, the phone will accept the call even if the called number is not belong to the phone. ¾ Auto Answer: If select it, the phone will auto answer when there is an incoming call. ¾ P2P IP Prefix: Set Prefix in peer to peer IP call. For example: what you want to dial is 192.168.1.
digit. for example, 4xxx expresses any number with prefix 4 which length is 4 will be forbidden to dialed out . Means matching any arbitrary number digit. For example, 6. expresses any number with prefix 6 will be forbidden to dialed out. User could make some device own IP, which is pre-specified, access to the MMI of the phone to config and manage the phone. Add or delete the IP address segments that access to the phone.
¾ First Codec：The fist preferential DSP codec: G.711A/u, G722, G.723, G.726-32, G.729. ¾ Second Codec ： The second preferential DSP codec: G.711A/u, G722, G.723, G.726-32, G.729. ¾ Third Codec ： The third preferential DSP codec: G.711A/u, G722, G.723, G.726-32, G.729. ¾ Forth Codec: The Forth preferential DSP codec: G.711A/u, G722, G.723, G.726-32, G.729. ¾ Fifth Codec: The fifth preferential DSP codec: G.711A/u, G722, G.723, G.726-32, G.729. ¾ Sixth Codec: The sixth preferential DSP codec: G.
This page is VPN setting page , the IP phone support the VPN with UDP and L2TP protocol .The parameters is as below. ¾ VPN IP： After VPN registered successfully, VPN server will give an IP aggress to the terminal. If there is a IP address shown on terminal (except for 0.0.0.0), it means your VPN has registered.
、 This functionality offers you more flexible dial rule, you can refer to the following content to know how to use this dial rule. When you want to dial an IP address, the entry of IP addresses is very cumbersome, but by this functionality, you can set number 179 to replace 192.168.1.179 here. When you want to dial a long distance call to China, you need dial an country code 86 before local phone number, but you can also dial number 0 instead of 86 after we make a setting according to this dial rule.
¾ Phone Number: The Number suit for this dial rule, can be set as full match or prefix match. Full match means that if the number user dialed is completely the same as this number, the call will use this dial-rule. Prefix match means that if prefix of the number that the user dials is the same as the prefix, the call will use this dial-rule, to distinguish from the full match case, you need to add “T” after the prefix number in the phone number setting. ¾ Call Mode: support SIP.
the phone number in the dial rule as 010T, and set the Alias as rep: 8610, and set the Del Length to 3. Then all calls begin with 010 will be changed to 8610 xxxxxxxx. ¾ Suffix (optional): Configure suffix, show no suffix if not set. Instance description as picture: ¾ 2T rule: if the call starts with 2, the first 2 will be deleted, and the rest number with be sent to IAX2 Server. ¾ 33 rule: Dial 33 and will send 83018618 to your server. Used as speed dial function.
As….” then you will save the config file in .txt format . ¾ Clear Config: user can restore factory default configuration and reboot the phone. If you login as Admin, the phone will reset all configurations and restore factory default; if you login as Guest, the phone will reset all configurations except for VoIP accounts (SIP1、SIP2 and IAX2) and version number.
192.168.1.1 or domain such as ftp.domain.com Meanwhile, it support sub directory such as 192.168.1.1/ftp/config/ or ftp.domain.com/ftp/config . ¾ Username: FTP user name (TFTP no need). ¾ Password: FTP password (TFTP no need). ¾ File name: the firmware or configuration file name that IP phone will search for in the server, if leave it as blank the IP phone with search the file with the name of its MAC such as 000102030405.
¾ Username: FTP server user name. ¾ Password: FTP server password. ¾ Config File Name: The name of configuration file. Normally users leave it as blank the IP phone search for the file with the name same as its MAC in the server. ¾ Config Encrypt Key: The encrypt key of confirmation file. ¾ Protocol Type: The protocol type that used for upgrading.:FTP,TFTP or Http. ¾ Update Interval Time: The interval time that the terminals search for new configuration file, counted in hour.
¾ Password: Set menu of keypad password, default is “123” ¾ Set KeyboardLock: The default password is “123”. It will take effect when you enable the keyboard lock. The default setting is unlock, if you press any key at this status, the system will remind you to input password ¾ Set Backlight Timeout: Set backlight time out, if IP Phone has not press any operation to active within the settings value, the backlight will off. ¾ Set Greeting Message: set the Greeting message on the LCD, default is VOIP PHONE.
¾ User Level：set new account level；root can read and change setting，general can only read ¾ Password：config password for new account ¾ Confirm：double confirm password If you want to make change on existing 】or 【 account, select the account an click 【 account can only modify or delete general account 】. General ¾ Keyboard Password：config password that you use keyboard to access the menu, must be in number.
¾ Ring Type: set different ring for different person If you want to make change on existing account, select the account an click 【Modify 】or 【Delete】. General account can only modify or delete general account Notice：Maximum records of phone book is 500pcs Multi line function is one of SIP line is busying, but other lines can get the call when have new calls with the line.
There are 9 function key on the phone, and be expanded to 29 with expander board.There are 5 types of the key: 1.NONE: do not use this key 2.Memory Key: set Number@Line/Subtype for the key,the number will be send out if you press the key. ¾ Number@Line/b: BLF (need server support). You can see the status of the blf number you set.When the key is green,means the number is free now. When the key turns red and blink,means the number is ringing.
3.Line: ¾ ¾ ¾ ¾ SIP1: SIP2: SIP3: SIP4: o use sip1 to call use sip2 to call use sip3 to call use sip4 to call IAX2: use iax2 to call 4.
whichadministrator can configure. This is a better way for log management.8 levels indebug information: Level 0---emergency: This is highest default debug info level. You system can not work. Level 1---alert: Your system has deadly problem. Level 2---critical: Your system has serious problem. Level 3---error: The error will affect your system working. Level 4---warning: There are some potential dangers. But your system can work.
¾ Server: type the IP address of time server ¾ Timezone: select correct time zone in list box ¾ Timeout: longest response time for SNTP ¾ Daylight Timeset: daylight setting through manual ¾ Manual Timeset: Time setting through manual ¾ Enable Daylight: Daylight saving time You can also set the time manually.
¾ Start Time：Display starts time of the outgoing record. ¾ Last Time：Display conversation time of the outgoing record. ¾ Called Number：Display the account/protocol/line of the outgoing record. Notice: It will cover existing automatically if the call log table has the new record.Call log will be cleared after phone reboot. ¾ Language Set: Set the language of phone, English is default. Because we use 14px font on LCD so the Chinese and Korean language are not supported but only can be supported on web.
Log out the configuration mode. If you want to re-configuration the phone, need to input the user and password to login again. Reboot IP phone, some settings need reboot to make it works. Please always save config before reboot, otherwise the settings will return to previous settings.
