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Safety Notices Please read the following safety notices before installing or using this phone. It is crucial for the safe and reliable operation of the device. Use the external power supply that is included in the package. Other power supplies may cause damage to the device, affect the telephone’s behavior or induce noise. Do not damage the power cord.
Table of Contents INTRODUCING E52 VOIP PHONE................7 ........................... 7 HANK YOU ........................7 ONTENTS ..........................7 EYPAD ......................8 NPUT UTPUT ORTS ......................8 NTRODUCTION LED I ......................9 NTRODUCTION 1.6.1 Power Indication LED (Power Light Enabled)..............9 1.6.2 Power Indication LED (Power Light Disabled)..............9 INITIAL CONNECTION AND SETTING..............
Introducing E52 VoIP Phone Thank you Thank you for purchasing the E52 Voice Over Internet Protocol (VoIP) telephone. The E52 is a fully featured telephone that provides voice communication over the data network. This phone has all the features of a traditional telephone and all gives access to many data service features.
Redial When off hook, this will dial the last called number. In stand-by mode, it will check the Outgoing Call. Speaker Activate speakerphone mode. phone Indicator This light blinks to indicate a missed call. light Various functions depending on the phone mode. Description will be shown in LCD.
Auto answer Contact DND(Do not Disturb) In hand free mode In handset mode Missed call Call forward LED Introduction 1.6.1 Power Indication LED (Power Light Enabled) LED Status Description Steady red Power on. Blinking red There is an incoming call. Power off.
WAN port on the back of the phone. Then use the Ethernet cable in the package to connect the LAN port on the back of the phone to the other device. The IP Phone now shares a network connection. 2. Connect the handset to the handset jack using the handset cable in the package. 3.
3. Press Enter. 4. Scroll down to “2 Advanced Settings.” 5. Press Enter. 6. The LCD will display “Enter Password”. 7. Input the password (default value is 123). 8. Press Eenter. 9. Scroll down to “2 Network.” 10. Press Enter. 11.
15. Press Down key. 16. Use the keypad to enter the Subnet Mask. 17. Press Save softkey. 18. Press Down key. 19. Use the keypad to enter the Gateway Address. 20. Press Save softkey. 21. Press Down key. 22. Use the keypad to enter the DNS 1 Address. 23.
23. Press Save softkey. 24. Press Back or Exit 6 times to return to idle screen. 25. Disconnect and reconnect the power supply so the phone will reboot and apply the new settings. Basic Functions Making a call 3.1.1 Call Device Calls can be made using two different devices: 1.
Call Forward This feature allows forwarding an incoming call to another phone number. The display shows icon. The following call forwarding events can be configured: Off: Call forwarding is deactivated by default. Always: Incoming calls are immediately forwarded. Busy: Incoming calls are immediately forwarded when the phone is busy. No Answer: Incoming calls are forwarded when the phone is not answered after a specific period.
complete the transfer. NOTE: Call waiting and call transfer must be enabled. NOTE: The SIP server must support RFC3515. 3.7.3 Semi-Attended Transfer During a conversation, press the XFER key, dial the number to which the call is to be transferred. Then press the Send softkey. When the third party phone begins to ring, press XFER to complete the transfer.
join by dialing a code plus the number for B or C. This assumes that B or C also support Join Call. The following chart shows how to configure this in the dial peer screen. *2* is the code. After saving the above configuration, A can dial *2* plus the number for B or C to join B and C’s call.
Speed dial This feature will allow you make speed dial easily. If you set up speed dial with name and tel numbers for 1~9, and then you can dial n# to make the corresponding speed dial number directly. Application 4.9.1 1.
4. Display will show “Ping IP Address” 5. After approximately 5 seconds, the display will show “OK” if the ping is successful or “Failed” is the ping is unsuccessful. Other Functions Call Forward If this feature is enabled, the phone will forward to another phone. 6.
Use vol-/vol+ to Enable. Ban Anonymous If this function is enabled, the phone will block calls with no Caller ID information. 1. Press Menu ->Features-> Enter->Ban Anonymous Call-> Enter. 2. Choose the SIP Account from which to Ban Anonymous Call. 3.
5.12 Auto Redial If Auto Redial is enabled, the phone will continue to retry a busy call. The user sets the retry interval and the number of times to redial. The user is also given the option to activate this feature on each busy call.
4. Use keypad to enter prefix. 5. Use Up/Down key to move to Length. 6. Use keypad to enter Length. 7. Use BACK or EXIT to return to idle screen. 5.17 Pre Dial If this feature is enabled, digits dialed on-hook will be transmitted when the phone goes off-hook Press Menu ->Features->...
Screen Settings 1. Press Menu ->Settings-> Enter->Basic Settings-> Enter->Screen Settings->Enter. 2. The following items can be set. Contrast – Set the contrast of the LCD. Contrast Calibration – Set the level of contrast that the current contrast setting provides. Backlight – Enable or disable LCD backlight. 3.
If Manual is chosen, the date and time must be entered. 3. Use Up/Down key to move to the following items. Use vol-/vol+ to make selection. SNTP Server – Time Server IP address – This is the only item that must be configured if auto is chosen.
6. SIP User – SIP User name 7. Auth User – User name for authentication 8. Auth Password – Password for authentication 7.1.2 Advanced Settings 1. Domain Realm – SIP Domain 2. Dial Without Registered – Enable or disable dialing with no SIP registration 3.
4. Auto Provision – Select DHCP Option, Plug and Play, or Phone Flash for autoprovision. 5. TR069 – Enable or disable configuration via TR069. 6. Backup – Select Config, Phonebook or none for backup. File name must be entered. 7. Upgrade – Select Image, MMI Set, BMF, Ring, Config, or Phonebook for upgrade. File name must be entered.
After entering the IP address, the following screen is displayed. After configuring the IP phone, remember to click SAVE under the Maintenance tab. If this is not done, the phone will lose the modifications when it is rebooted. Configuration via WEB 8.3.1 BASIC 8.3.1.1...
Network Shows the configuration information for WAN and LAN port, including connection mode of WAN port (Static, DHCP, PPPoE), MAC address, IP address of WAN port and LAN port, DHCP server status for LAN port (ENABLED or DISABLED). Accounts Shows the phone numbers and registration status for the 2 SIP LINES and 1 IAX2 server.
8.3.1.2.1 Static IP If Static IP is selected, this screen will be displayed. Information provided by the ISP should be entered. Click Back to return to the Wizard screen. Click Next to go to Quick SIP Settings 8.3.1.2.2 DHCP After selecting DHCP and clicking NEXT, the Quick SIP Settings screen will appear. Click Back to return to the Wizard screen.
8.3.1.2.4 Quick SIP Settings Field Name Explanation Display Name The name shown in caller ID. Server Address SIP server address either IP address or URI. Server Port SIP server port (usually 5060). Authentication User Login name or Authentication ID. Authentication Password SIP password.
8.3.1.3 Call Log Outgoing call logs can be seen on this page. Field Name Explanation Start Time Start time of the outgoing call Duration Duration of the outgoing call. Dialed Calls Account, protocol, and line of the outgoing call. 8.3.1.4 Language Field name Explanation...
8.3.2 Network 8.3.2.1 WAN Config Field Name Explanation Active IP Address The current IP address of the phone. Current Subnet Mask The current Subnet Mask. Current IP Gateway The current Gateway IP address. MAC Address The MAC address of the phone. MAC Timestamp Time the MAC address was obtained.
8.3.2.1.1 Static IP If Static IP is chosen, the screen below will appear. Enter values provided by the ISP. 8.3.2.1.2 DHCP If DHCP is chosen, all configuration information will be provided by a DHCP server. Contact the ISP to determine if DHCP is used. 8.3.2.1.3 PPPoE If PPPoE is chosen, the screen below will appear.
8.3.2.2 LAN Config Field Name Explanation IP Address LAN static IP. Subnet Mask LAN Subnet Mask. DHCP Service Activate DHCP server for LAN port. The phone must be rebooted for the DHCP server setting to take effect. Enable NAT operation Port Mirror Port Mirror can only be activated in bridge mode.
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Chart 1 shows a network switch with no VLAN. Any broadcast frames will be transmitted to all other ports. For example, and frames broadcast from Port 1 will be sent to Ports 2, 3, and 4. Chart 2 shows an example with two VLANs indicated by red and blue. In this example, frames broadcast from Port 1 will only go to Port 2 since Ports 3 and 4 are in a different VLAN.
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Field Name Explanation Enable LLDP Enable or Disable Link Layer Discovery Protocol (LLDP) Packet Interval The time interval for sending LLDP Packets Enable Learning Function Enables the telephone to synchronize its VLAN data with the Network Switch. The telephone will automatically synchronize DSCP, 802.1p, and VLAN ID values even if these values differ from those provided by the LLDP server.
8.3.2.4 Service Port Set the port values for Telnet/HTTP/RTP on this page. Field Name Explanation Web Server Type Specify Web Server Type – HTTP or HTTPS HTTP Port Port for web browser access. Default value is 80. To enhance security, change this from the default. Setting this port to 0 will disable HTTP access.
8.3.2.5 DHCP SERVICE Field Name Explanation DHCP Client Table IP-MAC mapping table. If the LAN port of the phone connects to a device, this table will show its IP and MAC address. Leased Table Name Name of the lease table. Start IP Address Beginning IP address of the lease table.
11. The size of lease table cannot be larger than the quantity of C network IP address. It is recommended to use the default lease table without modification 12. If the DHCP lease table is modified, the phone must be rebooted. 8.3.2.6 TIME&DATE Set the time zone and SNTP (Simple Network Time Protocol) server on this page.
Time Zone Local Time Zone Resync Period Time between resync to SNTP server. Default is 60 seconds. 12 -Hour Clock If checked, clock is 12 hour mode. If unchecked, 24 hour mode. Default is 24 hour mode. Date Format Specify the date format. Fourteen different formats are available. Date Separator Four date separators are available: /, - , .
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Field Name Explanation Choose the sip line to configured (SIP 1 – SIP2). Click the dropdown arrow to select the line. Status Shows registration status. Will show “Registered” if registered or “Unapplied” if not registered. Server Address SIP server IP address or URI. Server Port SIP server port.
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off hook. Hotline Number Number to be called in Hot Line Mode. Warm Line Wait Time Used in Hot Line Mode. Time the phone waits after off hook before dialing the hot line number. SIP Encryption Enable/Disable SIP Encryption. SIP Encryption Key SIP Encryption key.
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Code will disallow the phone to make anonymous calls. Ban Anonymous Off Allow Anonymous Calling function described above. In other Code words “Anonymous” will be transmitted for Caller ID. Keep Alive Type Specifies the NAT keep alive type. If SIP Option is selected, the phone will send SIP Option sip messages to the server every NAT Keep Alive Period.
address in via field. Enable GRUU Support for Globally Routable User-Agent URI (GRUU) Enable Displayname Puts quotation marks around the display-name in SIP messages. Quote For servers that require this. Enable user=phone Sets user=phone in SIP messages. For compatibility with servers that require this.
or “Unapplied” if not registered. Server Address IAX2 server address. Server Port IAX2 server port. Default is 4569. Account IAX2 account name for registration Password IAX2 registration password. Phone Number IAX2 phone number (usually the same as IAX2 account name). Local Port IAX2 local port.
Field Name Explanation STUN NAT Transversal Shows whether or not STUN NAT Transversal was successful. Server Address STUN Server IP address Server Port STUN Server Port – Default is 3478. Binding Period STUN blinding period – STUN packets are sent at this interval to keep the NAT mapping active.
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Example 3: Addition – Two examples are shown. In the first case, it is assumed that 0 must be dialed before any 11 digit number beginning with 13. In the second case, it is assumed that 0 must be dialed before any 11 digit number beginning with 135, 136, 137, 138, or 139. Two different special characters are used.
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enter the destination IP address or domain name. To use a dial rule on the SIP2 line, enter 0.0.0.2. Port Set the Signaling port, the default is 5060. Alias Set the Alias. This is the text to be added, replaced, or deleted. It is optional.
Set Phone Number, Alias and Dial “0106228” Delete Length. Phone number The SIP1 server will is XXXT and Alias is rep: xxx receive “86106228” If the dialed phone number starts with the digits in the Phone Number box, the matching digits will be replaced by the alias number.
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Fifth Codec The fifth codec choice G.711A,G.711A u, G.722, G.723.1, G.729AB, G.726-32, None Sixth codec The sixth codec choice G.711A,G.711A u, G.722, G.723.1, G.729AB, G.726-32,None Onhook Time Time the handset must be on hook to disconnect a call. Default is 200ms.
8.3.4.2 FEATURE This page configures various features such as Hotline, Call Transfer, Call Waiting, etc. Field Name Explanation DND (Do Not Disturb) DND might be disabled, phone for all SIP lines, or line for SIP individually. Enable Call Transfer If enabled, Call Transfer is allowed. Semi-Attended If enabled, Semi-Attended Transfer is allowed.
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Emergency Call The phone will dial the emergency call number even if the keyboard Number is locked. And multi numbers can be added by “,”, such as 911,999 Enable Password Dial When a number is entered beginning with the password prefix, the following N numbers after the password prefix will be displayed as *.
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call will be rejected. DND Return Code Specify SIP Code returned for DND. Default is 480 - Temporarily Not Available. Busy Return Code Specify SIP Code returned for Busy. Default is 486 – Busy Here. Reject Return Code Specify SIP Code returned for Rejected call. Default is 603 – Decline.
8.3.4.3 DIAL PLAN This phone supports 7 dialing modes: 17. Press "#" to Send– Dial the desired number, and press # to send it to the server. 18. Fixed Length – The number will be sent to the server after the specified number of digits are dialed.
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21. Press # to Do Blind Transfer - Press # after entering the target number for the transfer. The phone will transfer the current call to the third party. 22. Blind Transfer on Onhook - Hang up after entering the target number for the transfer. The phone will transfer the current call to the third party.
Note: End with “#”, Fixed Length, Time out and Digital Map Table can be used simultaneously. 8.3.4.4 CONTACT Enter the name, phone number and ring type for each contact here. Field Name Explanation Phonebook Tables Group Dropdown box to select group Name Contact name Office Number, Mobile...
Group Contact group for this contact Add Contact Name Contact name Office Number, Mobile Contact phone numbers Number, Other Number Line Select line for associated contact number Ring Type Ring type for this contact Group Setting Choose the group or groups for this contact and move them to the Selected list on the right.
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TFTP example for remote xml mode: Set the Phonebook Name as Fanvil - Server URL is tftp://192.168.1.3/admin/phonebook/index.xml. Remote Phonebook Settings Phonebook Name Phonebook name displayed on the phone. Server URL Server url of the remote phonebook. SIP Line SIP line for the remote phonebook. User/password Authentication username and password.
Mobile Display LDAP contact’s mobile phone number Other Display LDAP contact’s ohter number Display Name Allow display LDAP contact name or not 8.3.4.6 WEB DIAL This feature allows a call to be initiated by a computer. To place a call, enter the number in the Dial Number box, select the line in the Line Selection box and press the Dial button.
MCAST Settings Define the priority of the active call, 1 is the highest Prority priority, 10 is the lowest. The voice call in progress shall take precedence over all Enable Page Priority incoming paging calls. Name Listened multicast server name Host:port Listened multicast server’s multicast IP address and port.
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Auto Provision Settings Field Name Explanation Current Config Version Show the current config file’s version. If the version of configuration downloaded is higher than this, the configuration will be upgraded. If the endpoints confirm the configuration by the Digest method, the configuration will not be upgraded unless it differs from the current configuration.
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Plug and Play(pnp) Settings Enable PnP If this is enabled, the phone will send SIP SUBSCRIBE messages to a multicast address when it boots up. Any SIP server understanding that message will reply with a SIP NOTIFY message containing the Auto Provisioning Server URL where the phones can request their configuration.
ACS Server Type Select Common or CTC ACS Server Type. ACS Server URL ACS Server URL. ACS User User name for ACS. ACS Password ACS Password. TR069 Auto Login Enable/Disable TR069 Auto Login. "Inform" Sending Period Time between transmissions of “Inform” Unit is seconds. 8.3.6.2 Syslog Syslog is a protocol used to record log messages using a client/server mechanism.
Field Name Explanation Syslog Settings Server IP Syslog server IP address. Server Port Syslog server port. MGR Log Level Set the level of MGR log. SIP Log Level Set the level of SIP log. IAX2 Log Level Set the level of IAX2 log. Enable Syslog Enable or disable syslog.
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except for VoIP accounts (SIP1-2 and IAX2) and version number.
8.3.6.4 Update This page allows uploading configuration files to the phone. Update Field Name Explanation Web Update Browse to the config file, and press Update to load it to the phone. Web Update Various types of files can be loaded here including firmware, ring tones, local phonebook and config files in either text or xml format.
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Type Action to be executed by the phone. 1. Application update - download system update file 2. Config file export - Upload config file to FTP/TFTP server. It can then be named and saved. 3. Config file import - Download the config file from FTP/TFTP server.
8.3.6.5 Access User accounts can be added or deleted from this page. The authority of accounts can also be changed. Access Configuration Field Name Explanation LCD Menu Password Settings Menu Password Sets the password for entering the setup menu from the phone keypad.
General user can only read the configuration. Password Set the password Confirm Confirm the password User Management Select the account and click Modify to modify the selected account. Click Delete to delete the selected account. A General user can only add another General user. 8.3.6.6 Reboot Some configuration modifications require a reboot to become effective.
The Web filter is used to limit access to the phone. When the web filter is enabled, only the IP addresses between the start IP and end IP can access the phone. Field Name Explanation Start IP Address Beginning IP Address for MMI Filter End IP Address Ending IP Address for MMI Filter Add this filter range to the Web Filter Table...
Protocol Filter protocol type (TCP/ UDP/ ICMP/ IP) Port Range Set the filter Port range Src Address Set source address. It can be a single IP address or use * as a wild card. For example: 192.168.1.14 or *.*.*.14. Dest Address Set destination address.
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DMZ Configuration Servers in a network most vulnerable to attack are those which provide services to users outside the local network. Many times these computers are placed into their own sub-network to provide more protection to the rest of the local network. This sub-network is called a DMZ (taken from “demilitarized zone”).
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Application Layer Gateway (ALG) Settings Field Name Explanation IPSec ALG Enable/Disable IPSec encryption. Default is enabled. FTP ALG Allow the ALG to securely pass FTP traffic. Default is enabled. PPTP ALG Allow the ALG to securely pass PPTP traffic. Default is enabled. Network Address Translation (NAT) Table Shows the NAT TCP and UDP mapping tables NAT Table Option...
Inside IP Set the local IP address of device. Inside Port Set the LAN (inside) port for NAT mapping Outside Port Set the WAN (outside) port for NAT mapping Note: After entering settings, click the Add button to add new mapping table data. To delete an entry, enter its information and then click the Delete button.
8.3.7.5 Security Field Name Explanation Update Security File Select Security File Browse to the security file to be updated. Click the Update button to update. Delete Security File Select Security File Select the security file to be deleted. Click the Delete button to Delete.
Appendix Specification 9.1.1 Hardware Item Specification Power Adapter Input: 100-240V Output: 5V 1A Port 10/100Base- T RJ-45 1 PORT 10/100Base- T RJ-45 1 PORT Power Consumption Idle: 2.5W Active: 2.8W LCD Size 128x48 pixels Operation Temperature 0~40℃ Relative Humidity 10~65% Broadcom SDRAM 16MB...
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SIP support SIP domain SIP authentication none basic MD5 DNS Peer to Peer/ IP call Automatic line selection 9 Standard ring tones and 3 user-defined ring tones DTMF SIP info ...
Incoming Calls Outgoing Calls Missed Calls Max of 300 Records Each Supports vCard/XML/CSV Support IAX2 Programmable Soft Keys Code synchronization IP PBX IMS Supports Click to Dial via Web Phone Book ...
Digit-character map table Keypad Character Keypad Character 7 P Q R S p q r s 2 A B C a b c 8 T U V t u v 3 D E F d e f 9 W X Y Z w x y z 4 G H I g h i 5 J K L j k l 6 M N O m n o...
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