C58/C58P VoIP Phone User Manual Cortelco 1703 Sawyer Road Corinth, MS 38834 USA www.cortelco.com Tel: (662)287-5281 Fax:(662)287-3889 Version 1.
Safety Notices Please read the following safety notices before installing or using this phone. They are crucial for the safe and reliable operation of the device. Please use the external power supply that is included in the package. Other power supplies may cause damage to the device, affect the behavior or induce noise. Before using the external power supply, please be sure it is for use with your power voltage. Incorrect power voltage may cause fire and damage.
Table of Contents 1 INTRODUCING C58/C58P VOIP PHONE .................................................................... 7 1.1 THANK YOU .......................................................................................................................... 7 1.2 BOX CONTENTS .................................................................................................................... 7 1.3 KEYPAD ................................................................................................
.4 CLICK TO DIAL .................................................................................................................... 18 4.5 CALL BACK ......................................................................................................................... 18 4.6 AUTO ANSWER .................................................................................................................... 18 4.7 HOTLINE/WARMLINE .......................................................................
7.1 GENERAL............................................................................................................................ 25 7.2 ACCOUNT ........................................................................................................................... 25 7.2.1 Basic Settings .................................................................................................................... 25 7.2.2 Advanced Settings ..............................................................
8.3.4.5 REMOTE CONTACT ....................................................................................................... 60 8.3.4.6 WEB DIAL ........................................................................................................................ 60 8.3.5 Function Key ..................................................................................................................... 61 8.3.5.1 Function Keys ........................................................................
1 Introducing C58/C58P VoIP Phone 1.1 Thank you Thank you for purchasing the C58/C58P Voice Over Internet Protocol (VoIP) telephone. The C58/C58P is a fully featured telephone that provides voice communication over the data network. This phone has all the features of a traditional telephone and gives access to many data service features. The C58 and C58P are identical except that the C58P can be powered via Power over Ethernet (PoE).
heard. Volume -/+ Redial When off hook, this will dial the last called number. In stand-by mode, it will check the Outgoing Call. Speaker phone Indicator light Soft key 1/2/3/4 1.4 Port Adjust the volume by pressing these two keys. Activate speakerphone mode. This light blinks to indicate a missed call. Various functions depending on the phone mode. Description will be shown in LCD.
1.
1.6 LED Introduction 1.6.1 Programmable key LEDs for BLF LED Status Steady green Slow blinking red Steady red Fast blinking red Off 1.6.2 Programmable key LEDs for Presence LED Status Steady green Slow blinking red Steady red Fast blinking red Off 1.6.3 Description There are new voice mails. There is no new voice mail. Power Indication LED (Power Light Enabled) LED Status Steady red Blinking red Off 1.6.5 Description The object is online. The object is ringing. The object is active.
on the back of your phone to an Ethernet port. The following two figures show connection options. Direct network connection—This method requires at least one available Ethernet port. Connect the WAN port on the back of your phone to the Ethernet port. Since the phone has a built-in router, it can be connected directly to the network. Shared network connection—Use this method if you have a single Ethernet port which is already in use.
instructions below to change to either PPPoE or static IP. 2.2.1 1. 2. 3. 4. 5. 6. 7. 8. 9. 10. 11. 12. 13. 14. 15. 16. 17. 18. 19. 20. 21. 22. 23. 24. 25. 26. 27. 28. Press the MENU softkey. Scroll down to “3. Settings.” Press OK. Scroll down to “2. Advanced Settings.” Press OK. The LCD will display “INPUT PASSWORD”. Input the password (default value is 123). Press ENTER. Scroll down to “2. Network.” Press OK. Press OK to select WAN Settings. Scroll down to “4. PPPoE Settings.” Press OK.
10. 11. 12. 13. 14. 15. 16. 17. 18. 19. 20. 21. 22. 23. 24. 25. 26. 27. 28. 29. 30. 31. 32. 33. 34. Press OK. Press OK to select WAN Settings. Scroll down to “2. Static IP Settings.” Press OK. Use the keypad to enter the IP Address. Press SAVE softkey. Press DOWN ARROW. Use the keypad to enter the Subnet Mask. Press SAVE softkey. Press DOWN ARROW. Use the keypad to enter the Gateway Address. Press SAVE softkey. Press DOWN ARROW. Use the keypad to enter the DNS 1 Address. Press SAVE softkey.
17. 18. 19. 20. 21. 22. 23. 24. 25. Use LEFT ARROW or RIGHT ARROW to enable or disable DHCP Time. Press SAVE softkey. Press BACK softkey. Press UP ARROW or DOWN ARROW to scroll to “1. Connection Mode.” Press OK. Use LEFT ARROW or RIGHT ARROW to select “DHCP.” Press SAVE softkey. Press BACK or EXIT 6 times to return to idle screen. Disconnect and reconnect the power supply so the phone will reboot and apply the new settings. 3 Basic Functions 3.1 Making a call 3.1.
3.3 Do Not Disturb (DND) Press the DND softkey to active DND Mode. New incoming calls will be rejected and the display will show: icon. Press the DND softkey twice to deactivate DND mode. Incoming calls will be stored in the Call History. 3.4 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.
3.8 Mute When the Mute button is pressed during a conversation, the icon will be shown in the LCD. The distant party will not hear the party on the C58/C58P, but the distant party can still be heard. Press Mute again to return to normal conversation. 3.9 Call transfer 3.9.1 Blind Transfer During a conversation, press the XFER key, dial the number to which the call is to be transferred followed by "#" and then hang up. 3.9.
4. Press Send and wait for the other party to answer. 5. Use the arrow keys to select a call. 4 Advanced Functions 4.1 Call pickup This allows a third party to answer a call by dialing a code. For example: A calls B, but there is no answer. C can go off hook, dial a code plus B’s number, and pick up the call. The following chart shows how to configure this in the dial peer screen. *1* is the code. After saving the above configuration, C can dial *1* plus B’s phone number to pick up A’s call.
The user can select any prefix as long as it does not interfere with dialing rules. 4.4 Click to dial If User A browses to User B’s phone number or SIP address in the contact page and clicks it, User A’s phone will ring. After he goes off hook, the phone will call User B. Note：This feature requires that the PBX support click to dial. 4.5 Call back This function will redial the last received call. 4.
2. 3. 4. 5. 6. 7. Use the navigation keys to highlight the line for which you want to set voicemail. Press Edit Use the navigation keys to enable voicemail. Input the number. Press 2aB softkey if necessary to change the input method. Press Save to save the change. To hear a new voicemail, press the Voicemail softkey. Then press Dial. It may then be necessary to enter a password. 4.8.4 1. 2. 3. 4. 5. Ping Press Menu->Application->Ping->Enter. Enter the IP Address to be pinged.
None F_MWI – Message Waiting F_DND – Do Not Disturb F_HOLD – Hold F_B_TRANSFER – Blind Transfer F_PBOOK – Phonebook F_REDIAL – Redial F_PICKUP – Call Pickup F_JOIN – Join a call F_AUTOREDIAL – Auto Redial On F_UNAUTOREDIAL – Auto Redial Off F_CFWD – Call Forward F_CALLERS – Call List F_FLASH – Flash F_MEMO – Memo F_RELEASE – Release – Drop call F_LOCK – Locks the keypad.
3. Use LEFT ARROW or RIGHT ARROW to Enable. 4. Use UP ARROW or DOWN ARROW to access time setting. 5. Use keypad to enter time in seconds. 5.2 Auto Handdown This is the time after a call ends before the phone returns to the idle state. 1. Press Menu ->Features-> Enter->Auto Handdown-> Enter. 2. Use LEFT ARROW or RIGHT ARROW to Enable. 3. Use UP ARROW or DOWN ARROW to access time setting. 4. Use keypad to enter time in minutes. 5.
5.8 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. 1. Press Menu ->Features-> Enter->Auto Redial-> Enter. 2. Use LEFT ARROW or RIGHT ARROW to Enable. 3. Use UP ARROW or DOWN ARROW to select Interval and Times. 4. Press Save. 5.
be set. Example: A call is placed to 6625551212. Password is set to 662 and length is set to 3. Display will show 662***1212. 1. Press Menu ->Features-> Enter->Passwd Dial-> Enter. 2. Use LEFT ARROW or RIGHT ARROW to enable the feature. 3. Use UP ARROW or DOWN ARROW to move to Prefix. 4. Use keypad to enter prefix. 5. Use UP ARROW or DOWN ARROW to move to Length. 6. Use keypad to enter Length. 7. Use BACK or EXIT to return to idle screen. 5.
6.3 Ring Settings 6.3.1 Ring Volume 1. Press Menu ->Settings-> Enter->Basic Settings-> Enter->Ring Settings->Enter->Ring Volume->Enter. 2. Use LEFT ARROW or RIGHT ARROW to select the desired ring volume from the 9 choices. The phone will ring at the selected volume shortly after it is selected. 3. Press Save. 4. Use BACK or EXIT to return to idle screen. 6.3.2 Ring Type 1. Press Menu ->Settings-> Enter->Basic Settings-> Enter->Ring Settings->Enter->Ring Type->Enter. 2.
4. Press Save. 5. Use BACK or EXIT to return to idle screen. 6.6 Greeting Words This feature shows the words displayed in the upper left of the LCD. is VOIP PHONE. Default 1. Press Menu ->Settings-> Enter->Basic Settings-> Enter->Greeting Word->Enter. 2. Enter the message using the keypad. It may be necessary to change the input mode using the soft keys. Use DELETE to remove characters and 0 for space. Maximum message length is 12 characters. 3. Press Save. 4. Use BACK or EXIT to return to idle screen.
7.2.2 1. 2. 3. 4. 5. 6. 7. 8. 9. 10. 11. 12. Advanced Settings Domain Realm – SIP Domain Dial Without Registered – Enable or disable dialing with no SIP registration Anonymous – Privacy Support. Choose RFC3323, RFC3325 or None DTMF Mode – Choose RFC2833, SIP_Info, In-band, or Auto Use STUN – Enable or disable use of STUN Server. If enabled, the IP address of the STUN server must be entered. Local Port – Local SIP Port – Default 5060 Ring Type – Select ring type for this account. See Section 6.3.2.
name must be entered. 7.6 Factory Reset Choose Yes to return the phone to factory default settings. 8 Web Configuration 8.1 Introduction of configuration 8.1.1 Configuration Methods There are three methods which can be used to configure this phone: Phone keypad – As discussed in previous sections Web browser - Recommended way Telnet with CLI command 8.1.2 Password Configuration There are two levels of access: root level and general level.
After configuring the IP phone, remember to click SAVE under the Maintenance tab. is not done, the phone will lose the modifications when it is rebooted. 8.3 Configuration via WEB 8.3.1 BASIC 8.3.1.
Field Name Network Accounts Explanation 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). Shows the phone numbers and registration status for the 2 SIP LINES and 1 IAX2 server. 8.3.1.2 Wizard Select the appropriate network mode.
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. Click Next to go to the Summary screen. 8.3.1.2.3 PPPoE If PPPoE is selected, this screen will appear. Enter the information provided by the ISP.
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. SIP User Phone number. Enable Registration Submits registration information. Normally checked. Click Back to return to the IP Address screen. Click Next to see summary screen.
8.3.1.3 Call Log Outgoing call logs can be seen on this page. Field Name Start Time Duration Dialed Calls Explanation Start time of the outgoing call Duration of the outgoing call. Account, protocol, and line of the outgoing call. 8.3.1.4 Language Field name Language Greeting Words Explanation Set the language of phone. English is default. The greeting displayed on LCD when phone is idle. It has a maximum of 12 characters. Default is VOIP PHONE.
8.3.2 Network 8.3.2.1 WAN Config Field Name Active IP Address Current Subnet Mask Current IP Gateway MAC Address MAC Timestamp Explanation The current IP address of the phone. The current Subnet Mask. The current Gateway IP address. The MAC address of the phone. Time the MAC address was obtained. WAN Settings The phone supports three network modes. These are also discussed in Section 2.2. Static: Network parameters must be entered manually and will not change. All parameters are provided by the ISP.
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. Enter the information provided by the ISP.
8.3.2.2 LAN Config Field Name IP Address Subnet Mask DHCP Service Explanation LAN static IP. LAN Subnet Mask. Activate DHCP server for LAN port. The phone must be rebooted for the DHCP server setting to take effect. NAT Enable NAT operation Port Mirror Port Mirror can only be activated in bridge mode. If activated, the data stream from the WAN port is copied to the LAN port of the phone. Enable Bridge Mode If Bridge Mode is activated, the phone will not provide an IP address for the LAN port.
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. VLANs can be used to divide a network by restricting the transmission of broadcast frames.
Field Name Explanation Enable LLDP Packet Interval Enable or Disable Link Layer Discovery Protocol (LLDP) 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 Web Server Type HTTP Port HTTPS Port Telnet Port RTP Port Range Start RTP Port Quantity Explanation Specify Web Server Type – HTTP or HTTPS 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. Example: The IP address is 192.168.1.70 and the port value is 8090, the accessing address is http://192.168.1.70:8090.
8.3.2.5 DHCP SERVICE Field Name DHCP Client Table Leased Table Name Start IP Address End IP Address Explanation IP-MAC mapping table. If the LAN port of the phone connects to a device, this table will show its IP and MAC address. Name of the lease table. Beginning IP address of the lease table. Ending IP address of the lease table. A device connected to the LAN port will get an IP address between Start IP and End IP. Subnet Mask of the lease table. Network Gateway of the lease table.
8.3.2.6 TIME&DATE Set the time zone and SNTP (Simple Network Time Protocol) server on this page. Daylight savings time configuration and manual time and date entry are also done on this page Field Name Enable SNTP Enable DHCP Time Primary Server Secondary Server Time Zone Resync Period 12 -Hour Clock Date Format Date Separator Explanation Simple Network Time Protocol (SNTP) Settings Enable or Disable SNTP If this is enabled, phone will synchronize time with DHCP server.
Daylight Saving Time Settings Enable Enable daylight saving time. Offset(minutes) DST offset. Default is 60 minutes. Month Start and end month for DST Week Start and end week for DST Day Start and end day for DST Hour Start and end hour for DST Minute Start and end minute for DST Manual Time Settings Enter the values for the current year, month, day, hour and minute. All values are required. Note: Be sure to disable SNTP service before entering manual time and date. 8.3.3 VOIP 8.3.3.
Field Name Explanation Choose the sip line to configured (SIP 1 – SIP 2). 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. Default is 5060. Authentication User SIP account name (Login ID). Authentication Password SIP registration password. SIP User Phone number assigned by VoIP service provider.
Display Name Enable Registration Domain Realm Proxy Server Address Proxy Server Port Proxy User Proxy Password Backup Server Address Backup Server Port Server Name Set the display name. This name is shown on Caller ID. Check to submit registration information. SIP Domain if different than the SIP Registrar Server. SIP proxy server IP address or URI（This is normally the same as the SIP Registrar Server) SIP Proxy server port. Normally 5060. SIP Proxy server account. SIP Proxy server password.
Subscribe For MWI MWI Number Subscribe Period Conference Type Conference Number Registration Expires Enable Service Code DND On Code DND Off Code Always CFwd On Code Always CFwd Off Code Busy CFwd On Code Busy CFwd Off Code No Ans. CFwd On Code No Ans.
Local port Ring type Enable Rport Enable PRACK Enable Long Contact Convert URI Dial Without Registered Ban Anonymous Call Enable DNS SRV Enable Missed Call Log BLF List Number Enable BLF List Server Type RFC Protocol Edition Transport Protocol Anonymous Call Edition Keep Authentication Ans. With a Single Codec Auto TCP Enable Strict Proxy Enable GRUU Enable Displayname Quote Enable user=phone Click to Talk Strict Branch Enable Group Different VoIP Service providers may require different modes. SIP port.
Registration Failure Retry Time Registration failure retry time – If registration fails, the phone will attempt to register again after registration failure retry time. This will affect all lines 8.3.3.2 IAX2 Field Name Status Server Address Server Port Account Password Phone Number Local Port Voice Mail Number Voice Mail Text Echo Test Number Explanation Shows registration status. Will show “Registered” if registered or “Unapplied” if not registered. IAX2 server address. IAX2 server port.
Echo Test Text Refresh Time Enable Registration Enable G.729AB non- numeric, this number can be used to replace the echo test text. This allows dialing a number to perform an echo voice test. This function is provided to test whether communication through the server. Echo test text Expiration time of IAX2 server registration. Allowed values are between 60 and 3600 seconds. Enable/Disable IAX2 registration. Enable/Disable G.729 codec. 8.3.3.3 STUN Config STUN support is configured in this page.
Field Name STUN NAT Transversal Server Address Server Port Binding Period SIP Waiting Time SIP Line Using STUN Use STUN Explanation Shows whether or not STUN NAT Transversal was successful. STUN Server IP address STUN Server Port – Default is 3478. STUN blinding period – STUN packets are sent at this interval to keep the NAT mapping active. Waiting time for SIP. This will vary depending on the network.
Example 2: Substitution – To dial a long distance call to Beijing requires dialing area code 010 before the local phone number. Using this feature 1 can be substituted for 010. For example, to call 62213123 would only require dialing 162213123 instead of 01062213123. 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.
Field Name Phone number Destination Port Alias Explanation There are two types of matching: Full Matching or Prefix Matching. In Full matching, the entire phone number is entered and then mapped per the Dial Peer rules. In prefix matching, only part of the number is entered followed by T. The mapping with then take place whenever these digits are dialed. Prefix mode supports a maximum of 30 digits. Set Destination address. This is optional.
The phone will add the alias to the end of the dialed number if the dialed number matches the template in the Phone Number box. Set Phone Number, Alias and Delete Length. Phone number is XXXT and Alias is rep: xxx If the dialed phone number starts with the digits in the Phone Number box, the matching digits will be replaced by the alias number. If the dialed phone number starts with the digits in the Phone Number box, the phone will send out the dialed phone number and add the suffix number. 8.3.
Third Codec Fourth Codec Fifth Codec Sixth codec Onhook Time Default Ring Type Handset Input Volume Handset Output Volume Speakerphone Volume Ring Volume G729 Payload Length Tone Standard G722 Timestamps G723.1 Bit Rate Enable VAD DTMF Payload Type The third codec choice: G.711A/u, G.722, G.723, G.729, G.726, None The forth codec choice: G.711A/u, G.722, G.723, G.729, G.726, None The fifth codec choice G.711A/u, G.722, G.723, G.729, G.726, None The sixth codec choice G.711A/u, G.722, G.723, G.729, G.
8.3.4.2 FEATURE This page configures various features such as Hotline, Call Transfer, Call Waiting, etc. Field Name Do Not Disturb Enable Call Transfer Semi-Attended Transfer Enable Auto Handdown Auto Handdown Time Enable Auto Redial Auto Redial Interval Explanation If enabled, the phone will reject incoming calls. The callers receive busy tone. Outgoing calls may be made. If enabled, Call Transfer is allowed. If enabled, Semi-Attended Transfer is allowed.
Auto Redial Times Auto Headset Enable Intercom Enable Intercom Tone P2P IP Prefix Turn Off Power Light Emergency Call Number Enable Password Dial Password Dial Prefix Password Dial Length Ban Outgoing Enable Call Waiting Enable 3-way Conference Accept Any Call Enable Call Completion Enable Pre-Dial Enable Silent Mode Hide DTMF Maximum number of auto redial attempts. Automatically answers call on headset. If enabled, allows intercom calls.
Ring from Headset Enable Intercom Mute Enable Intercom Barge DND Return Code Busy Return Code Reject Return Code Active URI Limit IP Push XML Server Enable Call Waiting Tone Action URL Settings Block Out Settings If this is enabled and a headset is connected, ring tone will be played in the headset. If enabled, mutes incoming calls during an intercom call If enabled, the phone will auto-answer an intercom call during an outside call.
8.3.4.3 DIAL PLAN This phone supports 7 dialing modes: 1. End with “#”– Dial the desired number, and press # to send it to the server. 2. Fixed Length – The number will be sent to the server after the specified number of digits are dialed. 3. Time Out – Number will be sent to the server after the specified time. 4. User Defined – Customized rules created by the user. There is a special feature in the dial plan for the case where the user must dial an access code to get an external line.
5. 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. 6. 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. 7. Attended Transfer on Onhook - Hang up after the third party answers. The phone will transfer the current call to the third party. [] * .
8.3.4.4 CONTACT Enter the name, phone number and ring type for each contact here.
Office Number, Mobile Number, Other Number Ring Type Group Name Office Number, Mobile Number, Other Number Line Ring Type Group Setting Contact phone numbers Ring type for this contact Contact group for this contact Add Contact Contact name Contact phone numbers Select line for associated contact number Ring type for this contact Choose the group or groups for this contact and move them to the Selected list on the right.
8.3.4.5 REMOTE CONTACT Allows access to remote contact lists either via XML or LDAP. TFTP example: Set the Phonebook Name as cortelco - Server URL is tftp://192.168.1.3/admin/phonebook/index.xml. LDAP example: Server URL is ldap://192.168.1.3/dc= winline,dc=com. Remote Phonebook Setting Phonebook Name Phonebook name displayed on the phone. Server URL Server url of the remote phonebook. SIP Line SIP line for the remote phonebook. Authentication Authentication mode for remote phonebook.
8.3.5 Function Key The phone has 4 programmable DSS/Function keys with associated LEDs. The 4 directional arrow keys and the OK button are also programmable. This screen also sets the LCD contrast and enables the backlight. 8.3.5.1 Function Keys Field Name Contrast Enable Backlight Field Name Key Type Screen Configuration Explanation Set screen contrast Enable/disable LCD backlight. Function Key Settings Explanation Key Name Select the type of function the key is to perform.
Pickup Number Line – Seize a programmed line (SIP1/SIP2/IAX2) Memory Key – See Section 4.9.1. URL – Directly access a remote XML phonebook Parameters associated with the function. For example: The digits to be dialed by a key programmed for DTMF. Line on which the function is to be performed. Used with Key Event and Memory Key. Further specifies the type of function to perform.
8.3.6 Maintenance 8.3.6.1 Auto Provision The phone supports PnP, DHCP, and Phone Flash to obtain configuration parameters. They will be queried in the following order when the phone boots. DHCP PnP server Phone Flash Auto Provision Setting 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.
Encryption Key Save Autoprovision Information Save the Autoprovision username and password in the phone until the server url changes DHCP Option Settings Field Name DHCP Option Setting Custom DHCP Option Enable PnP PnP Server PnP Port PnP Transport PnP Interval Server Address Protocol Type Config File Name Update Interval Update Mode Explanation The phone supports configuration from Option 43, Option 66, or a Custom DHCP option. It may also be disabled. Custom option number. Must be from 128 to 254.
3. Update at time interval – update at periodic update interval TR069 Settings Enable TR069 ACS Server Type ACS Server URL ACS User ACS Password TR069 Auto Login "Inform" Sending Period Enable/Disable TR069 configuration Select Common or CTC ACS Server Type. ACS Server URL. User name for ACS. ACS Password. Enable/Disable TR069 Auto Login. 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.
Syslog Configuration Field Name Syslog Settings Server IP Server Port MGR Log Level SIP Log Level IAX2 Log Level Enable Syslog Web Capture Start Stop Explanation Syslog server IP address. Syslog server port. Set the level of MGR log. Set the level of SIP log. Set the level of IAX2 log. Enable or disable syslog. Capture a packet stream from the phone. This is normally used to troubleshoot problems.
8.3.6.3 Config Setting Config Setting Field Name Save Configuration Backup Configuration Clear Configuration Explanation Save the current phone configuration. Clicking this saves all configuration changes and makes them effective immediately. Save the phone configuration to a txt or xml file. Please note to Right click on the choice and then choose “Save Link As.” Logged in as Admin, this will restore factory default and remove all configuration information.
8.3.6.4 Update This page allows uploading configuration files to the phone. Update Field Name Web Update Explanation Web Update Browse to the config file, and press Update to load it to the phone. Various types of files can be loaded here including firmware, ring tones, local phonebook and config files in either text or xml format. TFTP/FTP Update Server Address FTP/TFTP server address for download/upload. The address can be IP address or Domain name with subdirectory.
Type Protocol 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. The configuration will be effective after the phone is reset. 4. Phone book export (.vcf, .csv, .xml) - Upload the phonebook file to FTP/TFTP server. It can then be named and saved. 5. PhoneBook import (.vcf, .csv, .
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. The password must be only digits. User Settings This table shows the current user accounts Add User User Set User Account name User Level There are two levels. Root user can modify the configuration.
8.3.6.6 Reboot Some configuration modifications require a reboot to become effective. button will cause the phone to reboot immediately. Note: Be sure to save the configuration before rebooting. 8.3.7 Clicking the Reboot Security 8.3.7.1 WEB FILTER WEB Filter 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.
8.3.7.2 Firewall Firewall Configuration Firewall rules can be used to prevent unauthorized Internet users from accessing private networks connected to this phone (input rule), or prevent unauthorized devices connected to this phone from accessing the Internet (output rule). Each rule type supports a maximum of 10 items.
When a connected device tries to access 192.168.1.118, the phone will deny the request because of the out_access rule. Access to any other IP address will be allowed. Click the Delete button to delete the selected rule. 8.3.7.3 Network Address Translation (NAT) NAT is the process of modifying IP address and port information in transition from a private to a public network. NAT allows the use of one public address to support many private addresses.
connectivity to specific hosts in the internal network, although communication with other hosts in the DMZ and to the external network is allowed. This allows hosts in the DMZ to provide services to both the internal and external network, while a firewall controls the traffic between the DMZ servers and the internal network clients. The following chart describes the network access control of DMZ.
Network Address Translation (NAT) Table Shows the NAT TCP and UDP mapping tables NAT Table Option Transfer Type Select the TCP or UDP protocol. 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. Notice: The phone supports 10M/100M adaptive.
VPN Server Address VPN User VPN Password Set VPN L2TP Server IP address. Set User Name access to VPN L2TP Server. Set Password access to VPN L2TP Server. 8.3.7.5 Security Field Name Select Security File Select Security File SIP TLS File HTTPS File OpenVPN Files 8.3.8 Explanation Update Security File Browse to the security file to be updated. Click the Update button to update. Delete Security File Select the security file to be deleted. Click the Delete button to Delete.
9 Appendix 9.1 Specification 9.1.1 Hardware Item Power Adapter Specification Operation Temperature Relative Humidity CPU SDRAM Input: 100-240V Output: 5V 1A 10/100Base- T RJ-45 1 PORT 10/100Base- T RJ-45 1 PORT RJ9 1 PORT Idle: 2.5W Active: 2.8W 128x48 74x28mm 0～40℃ 10～65% Broadcom 16MB Flash 4MB Dimension(L x W x H) Weight 295×295×175mm 1.5kg Port WAN LAN Headset Power Consumption LCD Size 9.1.2 Voice Features Supports 2 SIP servers Supports RFC3261 Codecs G.711A/u G.723.
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 DTMF Relay (In-Band) RFC2833 AUTO SIP applications Call Forward Call Transfer（Blind/Attended） Hold Call Waiting 3 Way Conference SMS Remote Pickup Join Call Redial Unredial Multi-line Intercom BLF Presence Push to talk Auto Redial Call Back Call control features Flexible dial plan Hotli
9.1.
Supports DHCP server on LAN QoS with DiffServ Network Tools in Telnet Server Ping Trace Route Telnet Client 9.1.
9.2 Digit-character map table Keypad Character Keypad 1@ Character 7PQRSpqrs 2ABC ab c 8TUVtuv 3DEFdef 9WXYZwxyz 4GHIghi */.
