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Ejointech ACOM214P

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2. eee None JE None JE None LL 1 Ep pe Bc Tp r1 a Ep on E None IL 1 None None FL None Lll won JE None on JE None Ll one AG None Ll on ILL E None LHl 8 3 5 3 SOFTKEY FUNCTION KEY EXT KEY SOFTKEY Softkey Settings Softkey Mode More Screen Call Dialer Unselected Softkeys Selected Softkeys None Delete Call Back CBack None Clear Dial History Exit In Join Missed MWI Next Line Next Out Pause Phonebook Dir Pickup Prev Line Prev Redial v m E EJ You can configure different functions in different screens for every softkey 8 3 6 Maintenance 8 3 6 1 Auto Provision i AUTO PROVISION SYSLOG UPDATE ACCESS REBOOT Auto Provision Settings Current Config Version 2 0002 Common Config Version 2 0002 CPE Serial Number 00100400XH0200100000000106e597052 User user Password Config Encryption Key Common Config Encryption Key Save Auto Provision Information DHCP Option Settings gt gt Plug and Play PnP Settings gt gt Phone Flash Settings gt gt TRO069 Settings gt gt DHCP Option Settings gt gt DHCP
3. number in this blank and then you need dial the phone number to realize calling to what the phone number is mapped In the prefix matching you need input your desired prefix number and T then dial the prefix and a phone number to realize calling to what your prefix number is mapped The prefix number supports at most 30 digits Destination Set Destination address This is optional config item If you want to set peer to peer call please input destination IP address or domain name If you want to use this dial rule on SIP2 line you need input 255 255 255 255 or 0 0 0 2 in it SIP3 into 0 0 0 3 Port Set the Signal port the default is 5060 for SIP Alias Set alias This is optional config item If you don t set Alias it will show no alias Note There are four types of aliases 1 Add xxx it means that you need dial xxx in front of phone number which will reduce dialing number length 2 All xxx it means that xxx will replace some phone number 3 Del It means that phone will delete the number with length appointed 4 Rep It means that phone will replace the number with length and number appointed You can refer to the following examples of different alias application to know more how to use different aliases and this dial rule Call Mode Select different signal protocol SIP or IAX2 Suffix Set suffix this is optional config item It will show no suffix if you don
4. 2 Attended Transfer During talk press the key Transf then input the number that you want to transfer to and press Send After that third party answers then press Transfer to complete the transfer You need enable call waiting and call transfer first If there are two calls you can just talk to one and keep hold to the other one The one who is keep hold cannot speak to you or hear from you In other way if user wants to invite the third party during the call they can press Conf to make calls mode in conference mode If user wants to stop conference user can press Split User must enable call waiting and three way call first Note the server that user uses must support RFC3515 or it might not be used 3 Alert Transfer During the talk press Transf firstly and then press Send after inputting the number that you want to transfer You are waiting for connection now press Transf and the transfer will be done To use this feature you need enable call waiting and call transfer first 3 9 Three Way conference call 1 Press the Conf softkey during an active call 2 The first call is placed on hold Then you will hear a dial tone Dial the number to conference in then press Send key 3 When the call is answered press Conf and add the first call to the conference 4 If you want to release the conference press Split Key Note You may need to start the phone call feature in order to achievethree way calling mode 3 10 Mul
5. 5 AUDIO FEATURE DIAL PLAN CONTACT REMOTE CONTA WEB DIAL MCAST MCAST Settings Priority 1 1 Enable Page Priority El Index Priority Name Host port N gt N Apply Use the multicast function to send notice to every member of the multicast is simple and easy By setting the multicast key on your phone you can send multicast RTP flow to the pre configured multicast address By listening multicast address is configured on the phone listen and play the multicast address to send the RTP stream Send multicast setting On the phone web page function key function key set a function key as shown DSS Key 8 Multicast v 239 1 1 1 1366 auro G7114 Value format IP Port the IP address of multicast is range from 224 0 0 0 to 239 255 255 255 port is greater than 1024 If multicast codec is G722 the LCD screen will displays HD which means the phone is sending high definition voice stream Operate steps 1 When the phone is idle press multicast key Multicast RTP stream is sended to pre configured multicast address IP Port The phone which listens to multicast address in the local network can receive the RTP stream Multicast functionkey LED lights yellow LCD screen displays the following 2 Press the hold softkey to hold the current multicast session 3 Press the end softkey again or multicast functionkey multicast session can be stopped
6. Application Layer Gateway ALG Settings IPSec ALG W FTP ALG PPTP ALG V Network Address Translation NAT Table Inside IP Address Inside TCP Port Outside TCP Port Inside IP Address Inside UDP Port Outside UDP Port NAT Table Option Transfer Type Outside Port Inside IP Address Inside Port Delete DMZ Settings NAT Configuration Field name explanation IPSec ALG It is an encryption technology Select it to enable IPSec ALG the default is enabled FTP is a service of connection layer which can FTP ALG transform intranet IP into extranet IP when intranet IP is sending out packet Select it to enable FTP ALG the default is enabled PPTP ALG Select it enable PPTP ALG the default is enabled Shows the NAT TCP mapping table Shows the NAT UDP mapping table Transfer Type Select the NAT mapping protocol style TCP or UDP Inside IP Set the IP address of device which is connected to LAN interface to do NAT mapping Inside Port Set the LAN port of the NAT mapping Outside Port Set the WAN port of the NAT mapping Notice After finish setting click the Add button to add new mapping table click the Delete button to delete the selected mapping table Shows the outside WAN port IP address and the inside LAN port IP address Notice 10M 100M adaptive means the network c
7. PRE teet 26 5 13 ACTION URL amp ACTIVE 26 6 THE BASIC SETTINGS OF 214 27 6 1 KEYBOARD 27 6 2 lt 9 E 27 6 3 RING y egal rtr PT 27 6 4 VOICE VOLUME c 27 6 5 m 28 6 6 GREETING WORDS cissicssccccscuiapcicesiessisucegendonsdeoniveascossessseaviaksegnsdcskibenisosacscies 28 6 7 LANGUAGE eR E 28 7 ADVANCED SETTINGS OF 214 29 v rc 29 P MIB Casio cm n 29 iit n ar e veu E QEVK REI 29 7 4 MAINTENANCE iccusssscaicasssdadsecuesetnscacdccsncsesabsssusdedasceucczansoeaaseonsseadeosasccsneseens 29 7 5 FACTORY RESET cscesscscsensseussssecsnssesevoncsencneussessceudscccetassessousedsssceistocenasscsssise 29 8 WEB CONFIGURATION ccccccsccsccsscccssccccccscccsccccscsccccscsccscssseee 20 8 1 INTRODUCTION OF CONFIGURATION ccccccccccccccssscccccccccccccccscccscccccscccccccccsscoosses 30 8 1 1 Ways to 30 8 1 2 Password Configurration cccccccoccccsscccscscccscscccccssccccscccccssccscssees 30 8 2 SETTING VIA WEB BROWSER ccccssssccsssssccccsccssccccsscssccccccssscccccsscsscccsessssccoeees 30 8
8. Unmute Missed Call IP Changed Idle To Busy Busy To Idle Block Out Settings Block Out ed E Active URI Specify the server IP that remote control phone for Limit IP corresponding operation Action URL Settings Action URL Specify the Action URL that Record the operation of phone Settings send this corresponding information to server url http InternalServer FileName xml Internal Server is server IP Filename is name of xml that contains the action message Block Out Settings Set Add Delete Limit List Please input the prefix of those phone numbers which you forbid the phone to dial out For example if you want to forbid those phones of 001 as prefix to be dialed out you need input 001 in the blank of limit list Block out and then you cannot dial out any phone number whose prefix is 001 X and are wildcard x means matching any single 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 Notice Black List and Limit List can record at most10 items respectively 8 3 4 3 DIAL PLAN This system supports 7 dial modes 1 End with dial your desired number and then press 2 Fixed Length the phone will intersect the number according to your specified length
9. configuration in the server is different with the running configuration CPE Serial Number Show CPE Serial Number User Specify FTP HTTP HTTPS server Username System will use anonymous if username keep blank Password Specify FTP HTTP HTTPS server Password Config Encrypt Input the Encrypt Key if the configuration file is Key encrypted Common Config Input the Common Encrypt Key if the Common Encrypt Key Configuration file is encrypted Save Save the username and password authentication Autoprovision message of http https ftp and input ID message in Information the phone until the url in the server changes DHCP Option Setting DHCP Option Setting Specify DHCP Option DHCP option supports DHCP custom option and DHCP option 66 and DHCP option 43 to obtain the parameters You could choose one method among them the default is DHCP option disable Custom DHCP A valid Custom DHCP Option is from 128 to 254 Option The Custom DHCP Option must be in accordance with the one defined in the DHCP server Plug and Play Enable PnP Enable PnP by selecting it than 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 PnP Server Specify the PnP Server PnP Port Specify the
10. t set it Delete Length Set delete length This is optional config item For example if the delete length is 3 the phone will delete the first 3 digits then send out the rest digits You can refer to examples of different alias application to know how to set delete length The following describes how to configure the number IP table to achieve the configuration of multiple accounts simultaneously 8T 0 0 0 2 5060 SIP del no suffix i 9T 0 0 0 0 5060 SIP del no suffix 1 9T means when you configure the SIP1 server and register then the user through all SIP1 call to dial a 9 before the number 8T means when you configure the SIP2 server and register then the user through all the numbers before calling SIP2 dial 8 0 0 0 0 4569 IAX2 le no suffi 2T means when you configure the IAX2 server and register then the user through all the IAX2 protocol number before the call can dial 2 Note For compatibility with 1 6 functions in the 1 7 version ofthe configuration file add Dialpeer With Line This field indicates whether to enable the on line inquiry function 0 is not enabled 1 means enabled The default is 0 Differences are as follows 1 Not enabled on line inquiry The function and the 1 6 version of the function is the same Type This rule indicates what protocol needs to go Destination indicates the destination address 0 0 0 1 represents go sip1 line 0 0 0 2 represents go sip2 line 0 0 0 x re
11. 1 Click Apply button after finished your setting IP Phone willsavethe setting automatically and new setting will take effect 2 If you modify the IP address the web wills not response by the old IP address Your need input new IP address in the address column to logon in the phone 3 If networks ID which is DHCP server distributed is same as network ID which is used by LAN of system system will use the DHCP IP to set WAN and modify LAN s networks ID for example system will change LAN IP from 192 168 10 1 to 192 168 11 1 when system uses DHCP client to get IP in startup If system uses DHCP client to get IP in running status and network ID is also same as LAN s system will refuse to accept the IP to configure WAN So WAN s active IP will be 0 0 0 0 8 3 22 LAN LAN QoS amp VLAN SERVICE PORT DHCP SERVICE TIME amp DATE i LAN Settings IP Address 192 168 10 23 Subnet Mask 255 255 255 0 DHCP Service v NAT v Port Mirror 4 Only works in the bridge mode Enable Bridge Mode Apply LAN Config Field name explanation LAN IP Address Specify LAN static IP Subnet Mask Specify LAN Netmask Select the DHCP server of LAN port or not After DHCP Service you modify the LAN IP address phone will amend and adjust the DHCP Lease Table and save the result amended automatically according to the IP address and Netmask You need reboot the phone and the DHCP server s
12. 5V AC 1A WAN 10 100M Connect it to Network jo LAN 10 100M Connect it to PC Expansion board Port type RJ 45 interface EXT Headset Port type RJ 9 connector LI Earpiece Port type RJ 9 connector 1 5 Icon introduction Icon Description Call out Key S gt Call in Call hold Auto answer Call mute Contact EM me DND Do not Disturb m lih In hand free mode In hook mode In headset mode SMS Missed call rig E Call forward 1 6 LED Status introduction Table 1 Programmable key LEDs for BLF LED Status Description Steady green The object is in idle status Slow blinking red The object is ringing Steady red The object is active Off The object is failed subscribe Off No subscribe Table 2 Programmable key LEDs for Presence LED Status Description Steady green The object is online Slow blinking red The object is ringing Steady red The object is active Off The object is failed No subscribe Table 3 Programmable key LEDs for line LED Status Description Steady green The account is active Fast Blinking green There is an incoming call to the account Slow Blinking green The call is on hold Slow Blinking red Registration is unsuccessful Off The line is not una
13. Call Established Call Terminated DND Enabled DND Disabled Always Forward Enabled Always Forward Disabled Busy Forward Enabled Busy Forward Disabled No Ans Forward Enabled Disabled x Ban Outgoing v Enable Call Waiting Enable 3 way Conference Accept Any Call second s Enable Call Completion Enable Pre Dial 10 1 180 second s Enable Silent Mode Hide DTMF Ring From Headset Enable Intercom Mute Enable Intercom Barge 10 1 100 zi DND Return Code Busy Return Code 110 Reject Return Code Active URI Limit IP PushXML Server 0 31 Enable Waiting Tone vj Enable Multi Line Enable Auto Switch Line Play Dialing DTMF Tone AUDIO emue DIAL PLAN CONTACT remore CONTAC WEB DIAL MCAST DR D SEIS Disabled 480 Temporarily Not Available 486 Busy Here 603 Decline 88 No Ans Forward Disabled Transfer Call Blind Transfer Call Attended Transfer Call Hold Resume Mute Unmute Missed Call IP Changed Idle To Busy Busy To Idle Block Out Settings Apply Block Out Add Te C Delete FEATURE Field name explanation Do Not Select DND the phone will reject any incoming call the Disturb callers will be reminded by busy
14. but any outgoing call from the phone will work well Ban If you select Ban Outgoing to enable it and you cannot dial Outgoing out any number Enable Call Transfer Enable Call Transfer by selecting it Semi Attend ed Transfer Enable Semi Attended Transfer by selecting it Enable Auto Redial Enable Auto Redial by selecting it then the phone reminds whether redial when the caller is busy or rejects Auto Redial interval Specify the Auto Redial interval Auto Redial Times Specify the Auto Redial interval Enable Call Enable Call Completion by selecting it Completion Enable Disable this feature in standby interface next number will Pre Dial realize the number rules send out over the time Enable the feature then the number will not be send out over the time Enable Call Enable Call Waiting by selecting it Then the phone reminds Waiting whether redial when the caller is busy or rejects if it s ok and the phone finds out that the caller is idle by sip message it will reminds whether redial Enable Enable 3 way conference by selecting it 3 way Conference Enable Call Disdale this function you will not hear the tone beep Waiting when there have multiple incoming calls Tone Accept Any If select it the phone will accept the call even if the called Call number is not belong to the phone Enable Auto The phone will hang up and r
15. 3 CONFIGURATION VIA WEB eese sse sn sense eese 31 8 3 1 BASIC 31 832 INET WORK 6 ii sscciscecisccccensscensccsstasecatsenccentsecesssevetessessdvostnoveneieneuchestusnte 36 AK MEA I 46 KE dguisl joe 62 SEUNCTION KEY Hae sisses eh 77 8 3 6 Mainten BI COs sssssisicssscssavesivosccncsvececevasieucconesdecsosescucovass sdeeiecoucseoveducssevtes 81 8 3 SECURITY isinsssvedsssodscderevacoscnastvabevcsguesensatsstackensusssaspscdcssetsnsaststecdascsvsasies 89 8 3 8 LOGOUT 96 SPECIFICATION saisssscssssecsdsscehsscsdccccussasssasndcdvessosacvesscasssosenavevonaseseussosdscsesouecoseaes 97 I yp Dio eb 97 9 1 2 Voice TEAtUT ES 97 9 1 3 Network featUreS sssssocsssssocssosoosssssoossoococssoosocssosoocssosoossossocsssssoessossosss 98 9 1 4 Maintenance and management sssesssocsssessosesooossosesosossooossosesosossosesososo 99 B 9 2 DIGIT CHARACTER TABLE sssscccococcesscccccoooooosececcoooooosocceocoooosssssooooo 99 1 Introducing ACOM214 VOIP Phone 1 1 Thank you for your purchasing ACOM214 Thank you for your purchasing C58 C58P C58 C58P is a full feature telephone that provides voice communication over the same data network t
16. Group Select the added groups then modify or delete and so on Name Input the name of the group then click the add button you can add a new group Ring Type Specify the ring type for the group as adding a new group Blacklist Settings Type Select the blacklist type you can select number or prefix of number Value Input number or prefix of number Line Select the sip line Notice the add button for adding a new blacklist the delete button for deleting one item the delete all button for deleting all items If user does not want to answer some phone calls add these phone numbers to the Black List and these calls will be rejected x and are wildcard x means matching any single digit For example 4xxx expresses any number with prefix 4 which length is 4 will be forbidden to be responded DOT means matching any arbitrary number digit For example 6 expresses any number with prefix 6 will be forbidden to be responded If user wants to allow a number or a series of number incoming he may add the number s to the list as the white list rule The configuration rule is number for example 123456 1234xx Black List 4119 Means any incoming number is forbidden except for 4119 Note End with DOT when set up the white list 8 3 4 5 REMOTE CONTACT AUDIO FEATURE DIAL PLAN CONTACT REMOTE CONTACT WEB DIAL MCAST Remote Phonebook Settings Index Phonebook Name Ser
17. MODE Choose PPPoE MODE click Next can config the PPPoE account password and SIP default SIP1 simply also can browse too Click Back can return to the last page Like Static IP MODE PPPoE Settings Service Name ANY User luser123 Password TT Back Nex Server name if PPPOE service providers are no special PPPOES requirements this name is usually the default value User Input your ADSL account Password Input your ADSL password Notice Click Finish button after finished your setting IP Phone will save the setting automatically and reboot After reboot you can dial by the SIP account 8 3 13 CALL LOG You can query all the outgoing through this page Call Information Start Time Duration Dialed Calls Call log Field name explanation Start Time Display the start time of the outgoing record Duration Display the conversation time of the outgoing record Dialed Calls Display the account protocol line of the outgoing record 8 3 1 4 LANGUAGE STATUS WIZARD CALL LOG LANGUAGE Language Language Selection English x Greeting Words Greeting Words VOIP PHONE LANGUAGE Field name Field name Language Set the language of phone English is default The greeting words will display on LCD when Greeting Words phone is idle It can support 12 chars the default chars are VOIP PHONE Notice the maximal length of the gree
18. Option Setting DHCP Option 66 Iz Custom DHCP Option 66 1285254 Plug and Play PnP Settings gt gt Enable PnP rl PnP Server 224 0 1 75 PnP Port 5060 PnP Transport UDP v PnP Interval 1 hours Phone Flash Settings gt gt Server Address 0 0 0 0 Config File Name Protocol Type FTP v Update Interval 1 hour s Update Mode Disabled Ejointech endpoint supports PnP and DHCP and Phone Flash to obtain the parameters The PnP and DHCP and Phone Flash are all deployed endpoint will go by the following process to try to obtain the server address and other parameters when it boots up DHCP option gt PnP server gt Phone Flash Auto Provision Field name explanation Auto Update Setting Current Config Version Show the current config file s version If the version of the configuration downloaded is higher than the version of the running configurations the auto provision would upgrade or stop here If the endpoints confirm the configuration by Digest method the endpoints wouldn t upgrade configuration unless the configuration in the server is different with the running configuration Common Config Version Show the common config file s version If the configuration downloaded and the running configurations are the same the auto provision would stop here If the endpoints confirm the configuration by Digest method the endpoints wouldn t upgrade configuration unless the
19. PnP Server PnP Transport Specify the PnP Transfer protocol PnP Interval Specify the Interval time unit is hour Phone Flash Server Address Set FTP TFTP HTTP server IP address for auto update The address can be IP address or Domain name with subdirectory Config File Name Set configuration file s name which need to update System will use MAC as config file name if config file name keep blank For example 000102030405 Protocol Type Specify the Protocol type FTP TFTP or HTTP Update Interval Specify update interval time unit is hour Update Mode Different update modes 1 Disable means no update 2 Update after reboot means update after reboot 3 Update at time interval means periodic update TRO69 Settings Enable 069 Enable TR069 by selecting it ACS Server Type Specify the ACS Server Type ACS Server URL Specify the ACS Server URL ACS User Specify ACS User ACS Password Specify ACS Password TRO069 Auto Login Enable TR069 Auto Login by selecting it Inform Sending Period Specify the inform Sending Period unit is second 8 3 6 2 SYSLOG Syslog is a protocol which is used to record the log messages with client server mechanism Syslog server receives the messages from clients and classifies them based on priority and type Then these messages will be written into log by some rules wh
20. The screen will show the current information and then press Del to delete Input your IP address Mask Gateway DNS and press Save to save what you input 5 Press Back six times to return to the idle screen 6 Check the status the screen shows Static the screen shows the IP address and gateway which were set just now if the phone could display the right time it shows that Static IP mode takes effect Setting DHCP mode 1 Press Menu Settings Advanced Settings then enter passwords and choose network gt WAN settings gt Connection Mode enter and choose DHCP through navigation keys and press the Save key 2 Press Back six times to return to the idle screen 3 Check the status the screen shows DHCP If the screen shows the IP address and gateway which were set just now it shows that DHCP mode takes effect 3 The basic function of 214 IP Phone 3 1 Making a call 3 1 1 Call Device You can make a phone call via the following devices 1 Pick up the handset icon will be showed in the idle screen 2 Press the Speaker button WM icon will be showed in the idle screen 3 Press the headset button if the headset is connected to the Headset Port in advance The icon 4 will be showed in the idle screen You can also dial the number first and choose the method you will use to speak to the other party 3 1 2 Call Methods You can press an available line button if there is more than one account
21. There are some options to configure Mode Date Time text Ring When the configuration is completed press Save 4 8 3 Voice Mail 1 Press Menu gt Application gt Voice Mail gt Enter 2 Use the navigation keys to highlight the line for which you want to set press Edit and use the navigation key to turn on the mode and the input the number Press 2aB softkey to choose the proper input method 3 Press Save to save the change 4 To view the new voicemail Press the Voicemail softkey directly Press Dial then you may be prompted to enter the password then you can listen to your new and old messages 4 8 4 Ping 1 Press Menu gt Ping gt Enter 2 Input the IP you want and press start key if input wrong you can press delete to modification the IP 3 After input the IP wait a moment it will display OK it meas ping successful or means ping failed 4 9 Programmable Key Configuration The phone has 4 programmable keys which are able to set up to many functions per key The following list shows the functions you can set on the programmable keys and provides a description for each function The default configuration for each key is N A which means the key hasn t been set for any functions 1 Set the type as Memory Key Press Menu gt Settings gt Basic Settings gt Enter gt Keyboard gt DSS Key Settings you have two options Select the line key and memory keys choose one you want to make the assignm
22. Time zone according to your location Resync Period Set the time out the default is 60 seconds 12 Hour Clock Switch the time mechanism between 12 hours and 24 hours Default is 24 hours mode Date format Specify the date format Daylight Saving Time Settings Enable Enable daylight saving time Offset minutes Setup the variety length Month Setup start and end month Week Setup start and end week Day Setup start and end day Hour Setup start and end hours Minute Setup start and end minutes Manual Time Settings Manual Time Settings Year Month Day Hour Minute Notice First of all you need to disable the SNTP service and above the date hours minutes each of which is required to complete and submit to make manual 8 3 3 VOIP 8 3 3 1 SIP Set your SIP server in the following interface SIP Line SIP 1 v Basic Settings gt gt Status Unapplied Domain Realm Server Address Proxy Server Address Server Port 5060 Proxy Server Port Authentication User Proxy User Authentication Password Proxy Password SIP User Backup Server Address Display Name Backup Server Port 5060 Enable Registration Server Name Codecs Settings gt gt Disabled Codecs Enabled Codecs G 711A G 711U G 722 G 7
23. and the server will turn off the function immediately No Answer CFwd On Code Set the No Answer CFwd On Code when you choose to enable the on answer forward function on your phone it will send message to the server and the server will turn on the function immediately When there are calls to the extension the server will forward it to the set number automatically based the forward type And the IP phone will not show the record in the call history anymore No Answer CFwd Off Code Set the No Answer CFwd Off Code when you choose to disable the busy forward function on your phone it will send message to the server and the server will turn off the function immediately Anonymous On Code Set the Anonymous On Code When you choose to enable the anonymous call function on your IP phone it will send information to the server and the server will enable the anonymous call function for your IP phone automatically Anonymous Off Code Set the Anonymous Off Code When you choose to disable the anonymous call function on your IP phone it will send information to the server and the server will disable the anonymous call function for your IP phone automatically Keep Alive Type Specify the keep alive type if the type is option the phone will send option sip message to server every NAT Keep Alive Period s then the server responses with 200 to keep alive If the type is UDP the pho
24. effect modify SIP call will use the modified port communication Sip Line Using STUN SIP Line Using STUN sipi e Use STUN Apply Choose line to set info about SIP There are 2 lines to choose Use STUN Enable Disable SIP STUN Notice SIP STUN is used to realize SIP penetration to NAT If your phone configures STUN Server IP and Port default is 3478 and enable SIP Stun you can use the ordinary SIP Server to realize penetration to NAT 8 3 3 4 DIAL PEER 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 156 to replace 192 168 1 119 here Dial Peer Table Number Destination Port Mode Alias Suffix Del Length 156 192 168 1 119 5060 SIP no alias no suffix 0 When you want dial a long distance call to Beijing you need dial area code 010 before local phone number but you can also dial number 1 instead of 010 after we make a setting according to this dial rule For example you want to dial 01062213123 but you need dial only 162213123 to realize your long distance call after you make this setting Dial Peer Table Number Destination Port Mode Alias Suffix Del Length 1T 0 0 0 0 5060 SIP rep 010 no suffix 1 To save the memory and avoid abundant input of user add the follow
25. functions Dial Peer Table Number Destination Port Mode Alias Suffix Deleted Length 13XXXXXXXXX 0 0 0 0 5060 SIP add 0 no suffix 0 13 5 9 0 0 0 0 5060 SIP add 0 no suffix 0 1 Increase in x matches any single digit for example If user makes the above configuration after user dials 11 digit numbers started with 13 the phone will send out 0 plus the dialed numbers automatically 2 Specifies a range that will match digit It may be a range a list of ranges separated by commas or a list of digits If user makes the above configuration after user dials 11 digit numbers started with from 135 to 139 the phone will send out 0 plus the dialed numbers automatically Use this phone you can realize dialing out via different lines without switch in web interface Dial Peer Table Number Destination Port Mode Alias Suffix Deleted Length 13XXXXXXXXX 0 0 0 0 5060 SIP add 0 no suffix 0 13 5 9 0 0 0 0 5060 SIP add 0 no suffix 0 156 192 168 1 119 5060 SIP no alias no suffix 0 TT 0 0 0 0 5060 SIP 010 no suffix 1 Add Dial Peer Phone Number Destination Optional Port Optional B Alias Optional Call Mode SIP wj Suffix Optional Deleted Length Optional Dial Peer Option 4 explanation Phone number There are two types of matching conditions one is full matching the other is prefix matching In the Full matching you need input your desired phone
26. page lists all its support of the action each action corresponds to a user defined URL When generating an action the phone is issued for the URL HTTP Get so as to achieve the purpose of reporting their actions 2 Active URI achieve results come from a functional understanding that the remote eg PC to send a URL to the phone the phone received will produce an action such as dial DND and so on Enter the phone web pages PHONE gt FEATURE enter the Active URL limit IP such as a PC IP Push XML Enter the web page of the phone gt PHONE gt FEATURE input Push XML Server e g PC IP then PC can push text SMS phonebook advertisement execute etc to phone to update the message or the phone makes an action 6 The Basic Settings of 214 6 1 Keyboard 1 Press Menu gt Settings gt Enter gt Basic Settings gt Enter gt Keyboard gt Enter 2 There are four items DSS Key settings Programmable Keys Desktop Long Pressed Soft Key You can set up respectively on them Press the key Enter to the interface then use the navigation keys to choose the function for the key according to you want 3 Press the key OK to save 6 2 Screen Settings 1 Press Menu gt Settings gt Enter gt Basic Settings gt Enter gt Screen Settings gt Enter 2 You can set Contrast Contrast Calibration and Backlight press Enter and use the navigation keys to set then press the key Save 6 3 Ring Settings 1 Press Menu gt Sett
27. same time MWI When the key is configured as MWI you are allowed to access voicemail quickly by pressing this key Call park 1 You need setting a server number when you have set what represent Call park If you have a calling and you busy now you could press the key and hear a number then you could choose other phone and input this number so you can directly recover call 2 Set the type as Line You can set these keys as line keys and press it it will enter dialer interface 3 Set the type as Key Event You can set these keys as Key Event and the subtype have many options Choose one and it will have corresponding function None MWI DND Hold Transfer Phone Book Redial Pick up Join Auto redial on Auto redial off Call Forward History Flash Memo Headset Release Press this button you can end all calls Lock You can select whether the keypad lock in the standby interface SMS Call Back Power Light Hide DTMF Agent Prefix Hot Desking You can clear the curren sip information and register their new sip information 4 Set the type as DTMF You can configure the key as DTMF This key function allows you to easily dial or edit dial number 5 Set the type as URL You need to match a XML Phonebook address pressing the button you can directly access the corresponding remote phonebook 6 Set the type as BLF List Key It needs the cooperation with the Broadsoft server The traditional BLF is t
28. t save the missed call log into the call history record and display the missed calls on the idle screen Click to talk Set click to Talk need practical software support Enable BLF List Enable BLF List by selecting it BLF list is a function which can monitor the group status it is not one to one monitoring but the information feedback from the server to decide which BLF list will monitor Use VPN Phone use vpn ip to communicate BLF List Number Specify the BLF List Number SIP Global Settings Strict Branch Enable the Strict Branch the value of the branch must be in the beginning of z9hG4k in via field of the invite sip message received or the phone won t response to the invite sip message Notice the deployment will become effective in all sip lines Enable Group Enable Group by selecting it then the phone enable the sip group backup function Notice the deployment will become effective in all sip lines Registration Failure Retry Time Specify the registration failure retry time if the phone register failed the phone will register again after registration failure retry time Notice the deployment will become effective in all sip lines 8 3 3 2 IAX2 IAX2 Status Unapplied Server Address Server Port 4569 Account Password mE Phone Number Local Po
29. th 1 Dial the number you want to call 2 Press History softkey use the navigation buttons to highlight your choice press Left Right button to choose Missed Calls Incoming Calls and Outgoing Calls 3 Press the R SEND button to call the last number called 4 Press the programmable keys which are set as speed dial button Then press the Send button or Dial softkey to make the call if necessary 3 2 Answering a call Answering an incoming call 1 Ifyou have no other line telephone lift the handset using or press the Speaker button Answer softkey to answer using the speaker phone or press the headset button to answer the headset 2 Ifyou are on a call currently press the answer softkey During the conversation you can alternate between Headset Handset and Speaker phone by pressing the corresponding buttons or picking up the handset 3 3 DND Press DND softkey to active DND Mode Further incoming calls will be rejected and the display shows BHO icon Press DND softkey twice to deactivate DND mode You can find the incoming call record in the Call History 3 4 Call Forward This feature allows you to forward an incoming call to another phone number The display showed 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 In
30. you to use the default lease table and not modify it 2 If you modify the DHCP lease table you need save the configuration and reboot 8 3 2 6 TIME amp DATE Setting time zone and SNTP Simple Network Time Protocol server according to your location you can also manually adjust date and time in this web page ac TR QoS amp VLAN SERVICE PORT DHCP SERVICE TIME amp DATE i Simple Network Time Protocol SNTP Settings Enable SNTP v Enable DHCP Time Primary Server 209 81 9 7 Secondary Server Timezone GMT 08 00 Beijing Chongging Hong Kong Urumai e Resync Period 60 second s 12 Hour Clock Date Format 1 Jan Mon Apply Daylight Saving Time Settings Enable Offset 60 minutes s Month March e October e Week B5 ss Day sunday v Sunday v 1 4 2 2 Manual Time Settings Year Month Day Hour Minute TIME amp DATE Field Name Explanation Simple Network Time Protocol SNTP Settings Enable SNTP Enable SNTP by selecting it Enable DHCP Time Enable DHCP Time by selecting it then the phone will automatically synchronize the standard time Primary Server Set SNTP Primary Server IP address Secondary Server Set SNTP Secondary Server IP address Time Zone Select the
31. 23 1 G 726 32 G 729AB t Advanced SIP Settings gt gt Forward Type Forward Number No Ans Fwd Wait Time Transfer Timeout SIP Encryption SIP Encryption Key RTP Encryption RTP Encryption Key Subscribe For MWI MWI Number Subscribe Period Enable Service Code DND On Code Always CFwd On Code Busy CFwd On Code No Ans CFwd On Code Anonymous On Code Keep Alive Type User Agent DTMF Type DTMF SIP INFO Mode Ring Type Enable Rport Enable PRACK Enable Long Contact Convert URI Dial Without Registered Ban Anonymous Call Enable DNS SRV Enable Missed Call Log SIP Global Settings gt gt Strict Branch Disabled v 0 second s 60 0 120 second s 3600 second s SIP Option 2833 10 11 Default Registration Failure Retry Time 32 Enable Hotline Hotline Number Warm Line Wait Time BLF Server Enable Auto Answer Auto Answer Timeout Enable Session Timer Session Timeout Conference Type Conference Number Registration Expires DND Off Code Always CFwd Off Code Busy CFwd Off Code No Ans CFwd Off Code Anonymous Off Code Keep Alive Interval Server Type RFC Protocol Edition Local Port Anonymous Call Edition Keep Authe
32. 3 Time Out After you stop dialing and waiting time out system will send the number collected 4 Press to Do Blind Transfer input the number you want to transfer to then press you can transfer the current call to the number 5 Blind Transfer on OnHook input the number you want to transfer to then hang up handle or press speaker you can transfer the current call to the number 6 Attend Transfer on OnHook hang up handle or press speaker you can realize the blind transfer function 7 Press the DSS key Blind Press dss key the current call will turn out blind 7 User defined you can customize digital map rules to make dialing more flexible It is realized by defining the prefix of phone number and number length of dialing In order to maintain the end user pbx secondary dial for dialing call mode When requested to enter a phone number prefix the sytem according to the rules in the closing number configuration rules re issue the dial tone the user continues to enter the number after the end of the closing number the phone number will be prefixed and analog secondary dial tone is sent to the back of the numbers together server For example In the list of rules in the configuration of the closing number 9 xxxxxxxx then when the user dials 9 the system to re play the dial tone dial the number the user to continue dial up is complete the phone is actually sent containing 9 9 numbers FEATURE DIAL P
33. ACOM214 VoIP Phone User Manual D 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 phone affect the behavior or induce noise Before using the external power supply in the package please check with home power voltage Inaccurate power voltage may cause fire and damage Please do not damage the power cord If power cord or plug is impaired do not use it it may cause fire or electric shock The plug socket combination must be accessible at all times because it serves as the main disconnecting device Do not drop knock or shake it Rough handling can break internal circuit boards Do not install the device in places where there is direct sunlight Also do not put the device on carpets or cushions It may cause fire or breakdown Avoid exposure the phone to high temperature below 0 C or high humidity Avoid wetting the unit with any liquid Do not attempt to open it Non expert handling of the device could damage it Consult your authorized dealer for help or else it may cause fire electric shock and breakdown Do not use harsh chemicals cleaning solvents or strong detergents to clean it Wipe it with a soft cloth that has been slightly dampened in a mild soap and water solution When lightnin
34. Back Next IP Address Input the IP address distributed to you Subnet Mask Input the Netmask distributed to you IP Gateway Input the Gateway address distributed to you Set DNS domain postfix When the domain which DNS Domain you input cannot be parsed phone will automatically add this domain to the end of the domain which you input before and parse it again Primary DNS Input your primary DNS server address Secondary DNS Input your standby DNS server address Quick SIP Settings Display Name Server Address Server Port Authentication User Authentication Password SIP User Enable Registration Next jJ Display Set the display name Server Address Input your SIP server address Server Port Set your SIP server port Authentication Input your SIP register account name User Authentication Input your SIP register password Password SIP User Input the phone number assigned by your VOIP service provider Enable Start to register or not by selecting it or not Registration WAN SIP Connection Mode Static IP Address IP Gateway Server Address Account Phone Number Registration Static IP 192 168 1 179 192 168 1 1 Disabled Finish Display detailed information that you manual config Choose DHCP MODE click Next can config SIP default SIP1 simply also can browse too Click Back can return to the last page Like Static IP
35. CK 19 ATO ANSWER M 19 4 8 APPLICATIONS ivisccesssovcnecscexscseseovscececonscnsseeuionsscoesosedsosccncceesssescaeconcssesdiscbenses 19 CB S MED UI r 19 4 8 2 1 OceiicccecihescctaseendcacsecnsocssencssiccessscdsdeoudeucbeasschesousisereceateisGeovdvecseesseass 20 4 8 3 V ice Ma thaiccccccccsccasassecsciacctenacessadccssecocecsnsetasccacsscecsegssesasacsaccescgnsasscaess 20 LEO MEM i E 20 4 9 PROGRAMMABLE KEY CONFIGURATION eeeeeeee eese eee osse eese esee eeee esses eese eoe 21 5 OTHER FUNCTIONS OF ACOM2414 24 5 1 AUTO HANDDOWN ee eenvec een ngeooecee thao cse Cb ve reU a Co repa oe eV een EET neni UE 24 5 2 BAN ANONYMOUS CALL ssssssesecssssssosososssacecssssscosesnssseseonsssnsosesicessoncatesnsesscnsense 24 5 3 DIAL PLAN Merc cs 24 5 4 DIAL PRER Pet 24 55 AUTO 24 5 6 CALL COMPLETION scsicsssscasssassssssseussosasscevesscsssenssceissetessonsossdss cniesussesssseeesosniss 25 5 7 RING FROM HEADSET yissssscsceviceistocccsssvscsavacasscasteenssesstecdbassucsscnascevapaadooviessees 25 5 8 POWER LIGHT e M 25 5 9 25 5 10 25 5 11 PASSWORD DIAL sissaccicsncesveviscerescesnesatssexssceesscarceoteseccsseupenehsseuesceusesevenevonies 26
36. G 729a b G 722 G 726 32 Support HD voice Echo cancellation G 168 Compliance in LEC additional acoustic echo cancellation AEC can reach 96ms max filter length in hands free mode Support Voice Gain Setting VAD CNG Support full duplex hands free SIP support SIP domain SIP authentication none basic MD5 DNS name of server Peer to Peer IP call Headphone interface RJ9 connector 2 line keys can be used with screen multi line operation or as a SIP line key 4 DSS keys Soft keys programmable function keys programmable Customizable multi language version the default is English SIP application support Call forward transfer blind transfer attended transfer Ringing Transfer Call hold call waiting conference call paging and intercom call park then grab interpolation Automatic Callback Click call auto secondary dial Flexible call control functions flexible dialing support hotline number calling reject reject blacklist certification calls white list barring do not disturb speakerphone automatic answer caller ID anonymous calls outgoing calls etc Support phonebook 500 records Incoming calls outgoing calls missed calls Each supports 300 records Support SMS Support MWI Support XML phonebook browser Support Speed dial Support SRTP BLF Code synchronization via IP PBX IMS Support click to dial via web phone book Voice codec setting for each SIP line Support keypad lock and em
37. LAN CONTACT REMOTE CONTA WEB DIAL MCAST Basic Settings v Press to Send Dial Fixed Length 11 to Send v Send after 5 second s 3 30 iv Press amp to Do Blind Transfer Blind Transfer on Onhook Attended Transfer on Onhook Press DSS Key to Do Blind Transfer Apply Dial Plan Table Plans L Add _ Delete DIAL PLAN Configuration Field name explanation Basic Setting Press to Send Set Enable Disable the phone ended with dial Dial Fixed Length Specify the Fixed Length of phone ending with Press to Do Blind Enable Blind Transfer On Hook when executing Transfer Blind Transfer End with press after inputting the number that you want to transfer the phone will transfer the current call to the third party Blind Transfer on Enable Blind Transfer on On Hook when executing OnHook Blind Transfer hang up after inputting the number that you want to transfer the phone will transfer the current call to the third party Attend Transferon Enable Attend Transfer on On Hook when OnHook executing Attended Transfer hang up after the third party answers the phone will transfer the current call to the third party Dial Plan Table Plans E Delete Below is user defined digital map rule Specifies a range that will match digit May be a range a list of ranges separated by commas or a list of digits Match any single digit that is diale
38. Lease Table Settings Leased Table Name Start IP Address End IP Address Leased Time minute s Subnet Mask IP Gateway DNS Server Address Add DHCP SERVICE Field name explanation IP MAC mapping table If the LAN port of the DHCP Lease Table phone connects to a device this table will show the IP and MAC address of this device Shows the DHCP Lease Table the unit of Lease time is Minute Lease Table Name Specify the name of the lease table Start IP Address Set the start IP address of the lease table Set the end IP address of the lease table the End IP Address network device connected to LAN port will get IP address between Start IP and End IP by DHCP Leased Time Set the Lease Time of the lease table Subnet Mask Set the Netmask of the lease table IP Gateway Set the Gateway of the lease table Set the default DNS server IP of the lease table DNS Click the Add button to submit and add this lease table DHCP Lease Table Delete Leased Table Name lan Delete Select name of lease table click the Delete button will delete the selected lease table from DHCP lease table DNS Relay Enable DNS Relay W Apply Select DNS Relay the default is enabled Click the Enable DNS Relay Apply button to become effective Notice 1 The size of lease table cannot be larger than the quantity of C network IP address We recommend
39. Next Line e Next Call None e Left None x None z Volume Down z None z Right e None e Volume Up Speed Dial OK Menu x None z None None z Apply Function Key Field name explanation Contrast Set contrast of screen Enable Backlight Set enable disable backlight Line Key Settings Line select Auto SIP1 SIP2 or IAX2 in function key type After you set it you pick up handset or hands free press this function key and then you can use the corresponding SIP line Function Key Settings key Show the function key s serial number Type Memory Key settings can be stored in key storage for each number the standby or off hook select the function keys on the keyboard can call this number Line set the dial mode Auto SIP1 SIP2 IAX2 Key Event functions monitor state DTMF In the call send DTMF URL You can input remote book url Value Set the type parameter values Line Choose which lines to use this feature Subtype Select the function parameters Key Event and Memory Event Pickup Number Please input the pickup number When SubType is BLF or presence NOTICE Memory keys can be configured through the following Speed Dial function through the configuration of the key corresponding to the number of ways as shown below DSS Key 1 Memory Key w 4111 siP1 1 Speed Dial User can press the F1 key to allocate th
40. Notice RTP stream is one side that is from a sender to a receiver when the phone initiates a multicast RTP session in a call the current call is on hold Receive multicast setting You can set up the phone monitoring 10 different multicast addresses to receive these multicast RTP stream You have two method to receive RTP stream of multicast that can be set up through the web page Enable priorities of normal calls and Enable page Priority Enable priorities of normal call by select it if the incoming RTP stream priority of multicast lower than the priority of current for normal calls the phone will ignore the RTP stream of multicast If the incoming RTP stream priority of multicast higher than the priority of current for normal calls the phone will receive the RTP stream of multicast and hold the current call Disabled priorities of normal call by select disable the phone will ignore all local network RTP stream of multicast Options as follows 1 10 the priority defined for normal calls 1 the highest level 10 the lowest level Disabled Ignore all RTP stream of multicast Enable Page Priority Page priority determines the phone how to handle the newly received multicast RTP stream when in a multicast session Enabled page priority the phone will automatically ignore the low priority multicast RTP stream and receive the high priority multicast RTP stream and hold the current multicast session If not enabled the phone wi
41. OM214 IP PHONE 14 3 1 MAKING A Weiner 14 3 11 Pall Device eerte 14 sm MENS ot nnn mec M 14 3 2 ANSWERING CALI cc csiscccussnctaverevetecsrervoceoseseeaderesebesorenssosoxsvecetesersvecersxiceses 14 33 D Foe prec mc Ln E MI PD ELM NP REL PN 15 3 4 CALL FORWARD ees esseeos eveeeps esesesckesssensdeseens eu vb vvdeuv e us eH eue np dee epe FREUE V ER 15 3 5 c 15 3 6 SCALE WAITING cies eee OE EP UP e UP HR FUR SEE RERO SE ER EEER ERE RE PEE ONE UE REV UE 15 37 a L H PTEE TRER 16 3 8 CALL TRANSFER oss scsssscccaisescesesescesssceissonctasseessduecensssntscoes nsvesereserescncuonesssenns 16 3 9 THREE WAY CONFERENCE CALL ccccssscccsssssccccsssssccccccsscccccscsssccccsscsscecsscsscees 16 3 10 MULTIPLE WAY CALL cecscssssccnicsecassescsnesstavesaccvesstenseevesencrensserstesocexeccsures 17 3 11 MULTILINE tr D 17 4 ADVANCED FUNCTION OF 214 18 4 1 4 2 4 3 4 4 4 5 4 6 CALL PIGKUPiisesisvsisessiensadsivsasstvssenstisivvasiiseseassistecadibnsseasbistesadabssscssbintosadabedss 18 JOIN CALL mee n 18 REDIAL UNREDIAL o ieeck ee essa obvio pU UU ao CoD bU SEES Ue RSKPRUE CONNU beo o 18 CLICK TO 19 CALL BA
42. SP codec G 711A u G 722 G 723 1 726 32 G 729AB None Onhook Time Specify the least reflection time of Hand down the default is 200ms Default Ring Type Set up the ring by default Handset Output Specify Output receiver Volume grade Volume Speakerphone Specify Speakerphone Volume grade volume G729AB Payload Set G729 Payload Length Length Tone Standard Select Tone Standard G722 Timestamps 160 20ms or 320 20ms is available G723 1 Bit Rate 5 3 kb s or 6 3 kb s is available Enable VAD Select it or not to enable or disable VAD If enable VAD G729 Payload length could not be set over 20ms DTMF Payload Set DTMF Payload Type Type 8 3 4 2 FEATURE In this web page you can configure Hotline Call Transfer Call Waiting 3 Ways Call Black List white list Limit List and so on Feature Settings DND Do Not Disturb Enable Call Transfer Semi Attended Transfer Enable Auto Handdown Auto Handdown Time Enable Auto Redial Auto Redial Interval 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 Length Enable Call History Enable Default Line Allow IP Call Play Talking DTMF Tone Action URL Settings Setup Completed Registration Success Registration Disabled Registration Failed Off Hook On Hook Incoming Call Outgoing Call
43. Start Click the start button when you need capture the WAN packet stream of the phone then open or save the file as the interface Stop Click the end button to stop capturing the packet stream 8 3 6 3 CONFIG AUTO PROVISION SYSLOG mr UPDATE ACCESS REBOOT Save Configuration Backup Configuration Clear Configuration Click Save button to save the configuration files Save all network and VOIP settings t cl Click Clear button to clear the configuration files Clear Config Setting Field name Explanation Save Configuration You can save all changes of configurations Click the Save button all changes of configuration will be saved and be effective immediately Backup Right clicks on Right click here and select Save Configuration Target As config File txt then you will save the config file in txt format or select Save Target As config File xml then you will save the config file in xml format Clear User can restore factory default configuration and Configuration 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 2 and IAX2 and version number 8 3 6 4 UPDATE You can update your configuration with your config file in this web page lal AUTO PROVISION SYSLOG CONFIG UPDATE ACCESS REBOOT
44. The following plans you can set Press to Send Timeout to Send Timeout Fixed Length Number Press to Do BXFER BXFER On Onhook AXFER On Onhook You can enable or disable each dial plan 5 4 Dial Peer 1 Press Menu gt Features gt Enter gt Dial Peer gt Enter 2 Press Add to enter the Edit interface and then input number and destination For example Number 1 Destination 1234 Then press Save 3 Input 1 number in the dial interface you can dial out 1234 You can refer to 8 3 3 4 5 5 Auto Redial 1 Press Menu gt Features gt Enter gt Auto Redial gt Enter 2 Choose Mode Enabled or Disabled through the navigation key If you choose Enable you also need to set Interval and Times and then press Save 3 After enable auto redial calling out someone if he is in busy it will pop up a prompt box whether to auto redial press OK the phone will call out him according the Interval and Times that you set 5 6 Call completion 1 Press Menu gt Features gt Enter gt Call Completion gt Enter 2 Enable the function through the navigation key and then Save 3 Call out others if he is in busy it will pop up a prompt Call Completion Waiting number Press OK when he is in idle it will pop up a prompt Call Completion Call number Press OK the phone will call out the number automatically 5 7 Ring From Headset 1 Press Menu gt Features gt Enter gt Ring From Headset gt Enter 2 Enable this funct
45. Web Update Select File z txt xml au vcf csv wav Update TFTP FTP Update Server Address User Password File Application Update Protocol FTP e Update Logo File Select File Delete Logo File Select File screensaver txt v Logo File screensaver txt 5779 Bytes Update Field name Explanation Web Update Click the browse button find out the config file Web Update saved before or provided by manufacturer download it to the phone directly press Update to save You can also update downloaded update file logo picture ring mmiset file by web TFTP FTP Update Server Address Set the FTP TFTP server address for download upload The address can be IP address or Domain name with subdirectory User Set the FTP server Username for download upload Password Set the FTP server password for download upload File name Set the name of update file or config file The default name is the MAC of the phone such as 000102030405 Notice You can modify the exported config file And you can also download config file which includes several modules that need to be imported For example you can download a config file just keep with SIP module After reboot other modules of system still use previous setting and are not lost Type Action type that system want to execute 1 Application
46. XFER New Call press OK enter the number then press Send and wait for the other party to answer When the multiple way calls you can press the arrow keys to select a call default supports 2 SIP account registration line and used simultaneously 4 Advanced function of ACOM214 4 1 Call pickup Call pickup is implemented by simulating pickup function of PBX it s that when A calls B B rings but no answer at this moment C can hook off and input an appointed prefix plus B s number pick up A s call and talk with A The following chart shows how to configure an appointed prefix in dial peer to have call pick up function Number Destination Port Mode Alias Suffix Deleted Length SP 0 0 0 0 5060 SIP rep pickup no suffix 3 1 means appointed prefix code After making the above configuration C can dial 1 plus B s phone number to pick up A s call User can set prefix in random in the case of no affecting current dialing rules 4 2 Join call When B is calling C A can join in the existing call by inputting an appointed prefix numbers plus B or C number if B or C also supports join call The following chart shows how to configure an appointed prefix in dial peer to have join call function Number Destination Port Mode Alias Suffix Deleted Length ar ha 0 0 0 0 5060 SIP rep joincall no suffix 3 2 means appointed prefix code After making the above configuration A can dial 2 plus B or C number to join B an
47. aled phone number adding suffix number When you dial 147 the SIP1 server will receive 1470011 8 3 4 PHONE 8 3 4 1 AUDIO In this page you can configure voice codec input output volume and so on AUDIO FEATURE Audio Settings DIAL PLAN CONTACT REMOTE WEB DIAL First Codec G 711A Second Codec 6 7110 Third Codec G 729AB Fourth Codec None Fifth Codec None Sixth Codec None Onhook Time 200 millisecond s Tone Standard China z Handset Volume 5 1 9 Default Ring Type Type 1 5 Speakerphone Volume 5 159 Headset Ring Volume 5 1 9 Headset Volume B 149 Speakerphone Ring Volume 5 149 G 729AB Payload Length 20ms G 723 1 Bit Rate 6 3kb s v G 722 Timestamps 160 20ms DTMF Payload Type 101 96 127 Enable VAD Enable MWI Tone Apply AUDIO Configuration Field name explanation First Codec The first preferential DSP codec G 711A u G 722 G 723 1 726 32 G 729AB None Second Codec The second preferential DSP codec G 711A u G 722 G 723 1 726 32 G 729AB None Third Codec The third preferential DSP codec G 711A u G 722 G 723 1 726 32 G 729AB None Fourth Codec The forth preferential DSP codec G 711A u G 722 G 723 1 726 32 G 729AB None Fifth Codec The fifth preferential DSP codec G 711A u G 722 G 723 1 726 32 G 729AB None Sixth codec The sixth preferential D
48. ard and other equipment physical consultations speed testing speed under bridge mode near to 100M in order to ensure the quality of voice and communications real time performance we made some sacrifices of NAT under the transmission performance Transmit with full capability only when system is idle so cannot guarantee that the transmission speed reach to 100M 8 3 7 4 VPN This web page provides us a safe connect mode by which we can make remote access to enterprise inner network from public network That is to say you can set it to connect public networks in different areas into inner network via special tunnel WEB FILTER FIREWALL NAT VPN SECURITY Virtual Private Network VPN Status IP Address 0 0 0 0 VPN Mode Enable VPN L2TP OpenVPN Layer 2 Tunneling Protocol L2TP VPN Server Address VPN User VPN Password VPN Configuration Field name explanation VPN IP Shows the current VPN IP address Select L2TP You can choose only one for current state After you select it you d better save configuration and reboot your phone Enable VPN Select it or not to enable or disable VPN VPN Server Set VPN L2TP Server IP address Address VPN User Set User Name access to VPN L2TP Server VPN Password Set Password access to VPN L2TP Server 8 3 7 5 SECURITY Update Security File Select Security File Browse Update Delete Securi
49. coming calls are forwarded when the phone is not answered after a specific period To configure Call Forward via Phone interface 1 Press Menu gt Features gt Enter gt Call Forwarding gt Enter 2 There are 4 options Disabled Always Busy and No Answer 3 Ifyou choose one of them except Disabled enter the phone number you want to forward your call to Press Save to save the changes 3 5 Call Hold Press the Hold button or Hold softkey to put your active call on hold 1 If there is only one call on hold press the hold softkey to retrieve the call 2 If there are more than one call on hold press the line button and the Up Down button to highlight the call then press the Unhold button to retrieve the call 3 6 Call Waiting 1 Press Menu gt Features gt Enter gt Call Waiting gt Enter 2 Use the navigation keys to active or inactive call waiting 3 Then press the Save to save the changes 3 7 Mute Press Mute button during the conversation icon will be showed in the LCD Then the called will not hear you but you can hear the called Press it again to get the phone to normal conversation 3 8 Call Transfer 1 Blind Transfer During talk press the key Transf and then dial the number that you want to transfer to and finished by Phone will transfer the current call to the third party After finishing transfer the call you talk to will be hanged up User cannot select SIP line when phone transfers call
50. cord When check the phone book record press this key again will return to idle mode Y Mute Press this key in calling mode you can hear the other side and the other side cannot hear you Once you press this key twice it would leave this mode and become normal Volume Turn down or turn up the volume by pressing these two keys REDIAL Redial 1 Inthe hook off hands free mode use the key to dial the last call number 2 In stand by mode it has a function to check the Outgoing Call 3 You could also find the specify contacts in phone book call records and use this number for quick dialing press this button you can dial quickly Hands free Make the phone into hands free mode Indicator The indicator would blinking if the IP Phone has light missed call Keys combination include functions such as History Directory DND Menu Del Redial Sen Soft key 1 2 3 4 d Quit Answer Divert Reject Hold Transfer Co nf Close and so on History View the Missed call Incoming Call and Outgoing Call LJ GJ DEE Digital 1 B 3 keyboard Inputting the phone number or DTMF GJE You can configure them the web page the DSS Keys configuration of each key to a different function 1 keys it would more convenient to you 1 4 Port for connecting Port Name Description Power swtich Input
51. d Match any arbitrary number of digits including none Tn Indicates an additional time out period before digits are sent of n seconds in length n is mandatory and can have a value of 0 to 9 seconds Tn must be the last 2 characters of a dial plan If Tn is not specified it is assumed to be TO by default on all dial plans Dial Plan Table Plans 1 8 xxx OXXXXXXX 911 4 9911x T4 Cause extensions 1000 8999 to be dialed immediately Cause 8 digit numbers started with 9 to be dialed immediately Cause 911 to be dialed immediately after it is entered Cause 99 to be dialed after 4 seconds Cause any number started with 9911 to be dialed 4 seconds after dialing ceases Notice End with Fixed Length Time out and Digital Map Table can be used simultaneously System will stop dialing and send number according to your set rules 8 3 4 4 CONTACT You can input the name phone number and select ring type for each name here Phonebook Table Group Al i 4 Hangup Index Name Office Number Mobile Number Other Number Ring Type Group Page Pre friend add J addtoBiacdist Delete Delete all Add Contact Name Ring Type Default e Office Number Line ato H Mobile Number Line auo e Other Number Line auo e Group Setting Unselected Selected friend home worl busi
52. d For example you set the password prefix is 3 enter the Password Length is 2 then you enter the number 34567 it will display 3 67 on the phone Password Specify the Password length Length DND Return Specify DND Return code Code Busy Return Code Specify Busy Return Code Reject Return Code Specify Reject Return Code Hide DTMF Specify the hide DTMF mode Push XML Specify the Push XML Server when phone receives request Server it will determine whether to display corresponding content on the phone which sent by the specified server or not Set Prefix in peer to peer IP call For example what you P2P IP want to dial is 192 168 1 119 If you define P2P IP Prefix as Prefix 192 168 1 you dial only 119 to reach 192 168 1 119 Default is If there is no Set it means to disable dialing IP Action URL Settings Setup Completed Registration Success Registration Disabled Registration Failed Off Hook On Hook Incoming Call Outgoing Call Call Established Call Terminated DND Enabled DND Disabled Always Forward Enabled Always Forward Disabled Busy Forward Enabled Busy Forward Disabled No Ans Forward Enabled No Ans Forward Disabled Transfer Call Blind Transfer Call Attended Transfer Call Hold Resume Mute
53. d C s call User can set prefix in random in the case of no affecting current dialing rules 4 3 Redial Unredial If B is in busy line when A calls B A will get notice busy please hang up If A want to connect B as soon as B is in idle he can use redial function at the moment and he can dials an appointed prefix number plus B s number to realize redial function What is redial function A can t not build a call with B when B is in busy then A will subscribe B s calling mode at 60 second intervals Once B is available A will get reminder of rings to hook off while a hooks off A will call B automatically If at this time A is occupied temporarily and unwilling to contact B A also can cancel the redial function by dialing an appointed prefix plus B s number before making the redial function Number Destination Port Mode Alias Suffix Deleted Length NET 0 0 0 0 5060 SIP rep redial no suffix 3 4 T 0 0 0 0 5060 SIP rep unredial no suffix 3 3 is appointed prefix code After making the above configuration A can dial 3 plus B s phone number to make the redial function 4 is appointed prefix code After configuration A can dial 4 to cancel redial function User can set prefix in random in the case of no affecting current dialing rules 4 4 Click to dial When user A browses in an appointed Web page user A can click to call user B via a link this link to user B then user A s phone will ring after A hoo
54. dband router ACOM214 have broadband routing capability as long as the ACOM214 properly connected to the WAN port on the broadband modem and connect your computer or other Internet capable devices connected to the ACOM214 s LAN port then you can use the phone s ability to connect to the Internet broadband routing The details setting mode please refer to 2 2 1 Network Settings Internet SD ADSL Cable Modem 2 1 2 Power adapter connection During this step please make sure your power connector match the power outlet meanwhile both voltage and electric current are also comply with the work phone 1 Plug power adapter to power socket 2 Plug power adaptor s DC output to the DC5V port of C58 C58P to start up 3 There will be displayed black line and INITIALIZING on the screen After finishing startup phone will show greeting current date and time and so forth 4 If phone has registered to the server you can place or answer calls 2 2 Basic Initialization ACOM214 is provided with a plenty of functions and parameters for configuration User needs some network and VoIP knowledge so that user could understand the meanings of parameters In order to make user use the phone more easily and convenient there are basic configurations introduced which is mandatory to ensure phone calls 2 2 1 Network Settings During setting network of the phone please make sure that network is connected already ACOM214 uses DHCP to
55. e Apply button So when devices execute to ping 192 168 1 118 system will deny the request to send icmp request to 192 168 1 118 for the out access rule But if devices ping other devices which network ID is 192 168 1 0 it will be normal Click the Delete button to delete the selected rule 8 3 7 3 NAT NAT is abbreviated from Net Address Translation it s a protocol responsible for IP address translation In other word it is responsible for transforming IP and port of private network to public also is the IP address mapping which we usually say Legal IP address uorje suei NAT Equipment Inner network d Private IP DMZ config In order to make some intranet equipment support better service for extranet and make internal network security more effectively these equipment open to extranet need be separated from the other equipment not open to extranet by the corresponding isolation method according to different demands We can provide the different security level protection in terms of the different resources by building a DMZ region which can provide the network level protection for the equipment environment reduce the risk which is caused by providing service to distrust customer and is the best position to put public information The following chart describes the network access control of DMZ DEZ area Inner Network area i WEB FILTER FIREWALL NAT mmm SECURITY
56. eneral This table shows the current user existed User Set account user name User Level Set user level Root user has the right to modify configuration General can only read Password Set the password Confirm Confirm the password Select the account and click the Modify to modify the selected account and click the Delete to delete the selected account General user only can add the user whose level is General 8 3 6 6 REBOOT AUTO PROVISION SYSLOG CONFIG UPDATE ACCESS REBOOT Reboot Phone Click Reboot button to restart the phone If you modified some configurations which need the phone s reboot to be effective you need click the Reboot then the phone will reboot immediately Notice Before reboot you need confirm that you have saved all configurations 8 3 7 SECURITY 8 3 7 1 WEB FILTER Web Filter Table Start IP Address End IP Address Option Web Filter Table Settings Start IP Address End IP Address Web Filter Setting Enable Web Filter Apply WEB Filter User could make some device own IP which is pre specified access to the MMI of the phone to config and manage the phone Field name explanation Web Filter Table Settings Add or delete the IP address segments that access to the phone Set initial IP address in the Start IP column Set end IP address in the End IP column and click Add to add this IP segment You can also c
57. ent use the navigation key to choose the type as memory key In the Dial field you have some options such as Normal Speed Dial Intercom BLF Presence MWI Call Park Speed dial You can configure the key as a simplified speed dial key This key function allows you to easily access your most dialed numbers Intercom You can configure the key for Intercom code and it is useful in an office environment as a quick access to connect to the operator or the secretary BLF BLF is also called Busy lamp field and it is used to prompt the user to pay attention to the state of the object than has been subscribed and used to cooperate with the server to pick up the phone call You can configure the key for Busy Lamp Field BLF which allows you to monitor the status idle ringing or busy of other SIP account User can dial out on a BLF configured key Please refer to LED Instruction for more detail about the LED status in different situation Note In the Web interface you can also set the pickup number to active the pickup function For example if you set the BLF number as 212 and the pickup number is 8 when there is an incoming call to 212 press the BLF key it will call out the 189 automatically to pickup the incoming call on 212 Presence Presence is called present and compared to the BLF it can also check whether object online Note You can not subscribe the BLF and presence station of the same number at the
58. ergency call during the keypad lock Customized lcd logo Headset speakerphone Ringing Selection Ringing tone custom configuration parameters Group listening 9 1 3 Network features WAN LAN support bridge and router model Support basic NAT and NAPT Support PPPoE for xDSL Support VLAN optional voice vlan data vlan NAT Penetrate Stun Penetrate Support DMZ Support VPN L2TP OPEN VPN function Wan Port supports main DNS and secondary DNS server can select dynamically to get DNS in DHCP mode or statically set DNS address Support DHCP client on WAN Support DHCP server on LAN QoS with DiffServ Network tools in telnet server including ping trace route telnet client 9 1 4 Maintenance and management Upgrade firmware through POST mode Web telnet and keypad management Management with different account right LCD and WEB configuration can be modified into requested language and support multi language dynamically shifted Upgrade firmware through HTTP FTP TFTP Telnet remote management upload download setting file Support Syslog Support Auto Provisioning upgrade firmware or configuration file 9 2 Digit character map table Keypad Character Keypad Character 1 10 7 7PQRSpqrs a Y 2ABCabc f 7 2 8 8TUVtuv ABC TUV AS Z f li 3DEFdef 3 9 OWXYZwXyz DEF WXYZ 4 4GHIghi
59. etting will take effect NAT Select NAT or not Select Port Mirror or not it only works in bridge mode the function of the port mirror is that copy Port Mirror the data stream from the WAN port to the LAN port of the phone Select Bridge Mode or not If you select Bridge Enable Bridge Mode the phone will no longer set IP address for Mode LAN physical port LAN and WAN will join in the same network Click Apply the phone will reboot Notice When LAN IP or bridge mode status is changed the system will reboot If you choose the bridge mode the LAN configuration will be disabled 8323 QoS amp VLAN The VOIP phone support 802 1Q P protocol and DiffServ configuration VLAN functionality can use different VLAN IDs by setting voice VLAN and data VLAN The VLAN application of this phone is very flexible Do not use VLAN After Switchboard received the Broadcast Frame transmit to every other port except the send port Switchboard Broadcast Frame Use VLAN After Switchboard received Switchboard the Broadcast Frame only transmit it to other port which belong to same VLAN with send port Broadcast Broadcast Frame Frame x VLAN EY 2 Q Broadcast Qi Domain SU CU Domain Chart 2 In chart 1 there is a layer 2 that switches without setting VLAN Any broadcast frame will be transmitted to the other ports except the send port For example a broadcast informati
60. eturn to the idle automatically Hand down at hands free mode Auto Hand Specify Auto Hand down Time the phone will hang up and down Time return to the idle automatically after Auto Hand down Time at hands free mode and play dial tone Auto Hand down Time at handset mode Ring From Enable Ring From Handset by selecting it the phone plays Headset ring tone from handset Enable Enable Intercom Mode by selecting it Intercom Enable Enable mute mode during the intercom call Intercom Mute Enable If the incoming call is intercom call the phone plays the Intercom intercom tone Tone Enable Enable Intercom Barge by selecting it the phone auto Intercom answers the intercom call during a call If the current call is Barge intercom call the phone will reject the second intercom call Enable Enable Silent Mode by selecting it the phone light will red Silent Mode blink to remind that there is a missed call instead of playing ring tone Turn Off Power Light Enable Turn Off Power Light by selecting it Emergency Call Number Specify the Emergency Call Number Despite the keyboard is locked you can dial the emergency call number Enable Enable Password Dial by selecting it When number entered Password is beginning with the password prefix the following N Dial numbers After the password prefix will be hidden as N stand for the value which you enter in the Password Length fiel
61. g do not touch power plug or phone line it may cause an electric shock Do not install this phone in an ill ventilated place You are in a situation that could cause bodily injury Before you work on any equipment be aware of the hazards involved with electrical circuitry and be familiar with standard practices for preventing accidents Table of Content 1 INTRODUCING ACOM214 VOIP 1 1 1 2 1 3 1 4 1 5 1 6 THANK YOU FOR YOUR PURCHASING 214 6 DELIVERY CONTENT ss isiicoisscassecedepvasasnabsceavcsgsncascesedassenseasessneebsenssvossssenvseeneeess 6 aeo 7 PORT FOR CONNECTING 5 cisccsissdsecosccsceceescesececscsasvesecsiesocecsascosessuonecsesivenssscsees 8 ICON Bi wucti or P opu o nt M 8 LED STATUS INTRODUCTION sissscsssissscesssececnsstesnsecesnescesssssesnesbesieessonsesosvencacese 9 2 INITIAL CONNECTING AND SETTINGS e eeeeeeeeees 11 2 1 CONNECT THE POWER AND NETWORK eeeeees osse eee sos see esos sse e sees osse ee eosseeno 11 2 1 1 Connect to neDWorIo bospecke evo ioc tps Ve ise epe bes Pepi n 11 2 1 2 Power adapter connection eere eee eee eee ee eene eee en esee eo sse eos seno 12 2 2 BASIC INITIALIZATION 12 2 2 1 Network Settings se sssessooscsosossooosoooesosossoocsosossosesosossooossosesosossosesososo 12 3 THE BASIC FUNCTION OF AC
62. get WAN IP configurations so phone could access to network as long as there is DHCP server in it If there is no DHCP server available phone has to be changed WAN network setting to Static IP or PPPoE Setting PPPOE mode For ADSL connection 1 Get PPPoE account and password first 2 Press Menu gt Settings gt Advanced Settings then enter passwords and choose network gt WAN settings gt Connection Mode enter and choose PPPoE through navigation keys and press the Save key 3 Press Back then choose PPPoE Set press Enter 4 The screen will show the current information Press Del to delete it then input your PPPoE user and password and press Save 5 Press Back six times to return to the idle screen 6 Check the status If the screen shows Negotiating it shows that the phone is trying to access to the PPPoE Server if it shows an IP address then the phone has already get IP with PPPoE Setting Static IP mode Static ADSL Cable or no PPPOE DHCP network 1 Prepare the network s parameters first such as IP Address Net mask Default Gateway and DNS server IP address If you don t know this information please contact the service provider or technician of network 2 Press Menu gt Settings gt Advanced Settings then enter passwords and choose network gt WAN settings gt Connection Mode enter and choose Static through navigation keys and press the Save key 3 Press Back then choose Static Set press Enter 4
63. hat every number will need to be subscribed so if the numbers that subscribed is so many that it will cause to obstruction However BLF List Key will put the numbers that needed to be subscribed in a group and the phone use the URL of the group to subscribe and analyze the specific information of each number such as number name state and so on according to the notifications from the server Then set the idle Memory key as BLF List Key later if the state of an object changes the corresponding LED will change 7 Set the type as Multicast Set the multicast address and speech coding press this key to initiate the multicast Note More features see 8 3 4 7 5 Other functions of 214 5 1 Auto Handdown 1 Press Menu gt Features gt Enter gt Auto Handdown gt Enter 2 Set the Mode Enable through the navigation key then set Time unit is minute then press Save 3 When the call ends after the time that you have set the phone will back to the idle interface 5 2 Ban Anonymous Call 1 Press Menu gt Features gt Enter gt Ban Anonymous Call gt Enter 2 Choose which sip you want to enable Ban Anonymous Call and then press Enter choose Enabled or Disabled through navigation key 3 If you choose Enabled the others can t call the phone by anonymous If you choose Disabled the others can call the phone by anonymous 5 3 Dial Plan 1 Press Menu gt Features gt Enter gt Dial Plan gt Enter 2
64. hat your computer uses This phone s functions not only much like a traditional phone allowing to place and receive calls and enjoy other features that traditional phone has but it also own many data services features which you could not expect from a traditional telephone This guide will help you easily use the various features and services available on your phone 1 2 Delivery Content Please check whether the delivery contains the following parts The base unit with display and keypad The handset The handset cable The Ethernet cable The power supply Attentions The ACOM214 may cause damage if you do not use a power adapter with ACOM214 Power adapter specifications due to different areas or differentiated shipments if the product supplied power adapter can not be used locally please consult your local dealer The user manual you may download from our website Here is the appearance of IP Phone description How onog o EDO 2 8 3 pud AOE 1 3 Keypad Function Description Navigation Navigation key assist users for operating In desktop dialer calling desktop long pressed state they have special function You can configure through the web page according to your patterns of use DIR C Phone book Access to phone book check the record list and add new records and revise the re
65. ich administrator can configure This is a better way for log management 8 levels in debug information Level 0 emergency This is highest default debug info level You system cannot 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 Level 5 notice Your system works well in special condition but you need to check its working environment and parameter Level 6 info the daily debugging info Level 7 debug the lowest debug info Professional debugging info from R amp D person At present the lowest level of debug information is info debug level only can be displayed on telnet i AUTO PROVISION SYSLOG CONFIG UPDATE ACCESS REBOOT Syslog Settings Server Address 0 0 0 0 Server Port 514 MGR Log Level None e SIP Log Level None IAX2 Log Level None Enable Syslog Web Capture Stop Syslog Configuration Field name explanation Syslog Setting Server Address Set Syslog server IP address Server Port Set Syslog server port MGR Log Level Set the level of MGR log SIP Log Level Set the level of SIP log 2 Log Level Set the level of IAX2 log Enable Syslog Select it or not to enable or disable syslog Web Capture
66. ings gt Enter gt Basic Settings gt Enter gt Ring Settings gt Enter 2 You can set Ring Volume and Ring Type press Enter and use the navigation keys to set then press the key Save In the Ring Type the default system rings have nine and the custom ringtones have three that can be set through the web page 6 4 Voice Volume 1 Press Menu gt Settings gt Enter gt Basic Setting gt Enter gt Voice Volume gt Enter 2 Use the navigation keys to turn down or turn up the voice volume then press the key Save 6 5 Time amp Date 1 Press Menu gt Settings gt Enter gt Basic Settings gt Enter gt Time amp Date gt Enter 2 You have two options to choose Auto and Manual use the navigation keys to choose then press Save 6 6 Greeting Words 1 Press Menu gt Settings gt Enter gt Basic Settings gt Enter gt Greeting Words gt Enter 2 You can enter the message and press Save it will display in the phone screen when the phone start up 6 7 Language 1 Press Menu gt Settings gt Enter gt Basic Settings gt Enter gt Language gt Enter 2 ACOM214 support three languages you can use the navigation keys to choose The default two languages are English and Chinese 7 Advanced Settings of 214 7 1 Accounts Press Menu gt Enter gt Advanced settings and then input the password to enter the interface the default password is 123 You can set it through the web page Then choose Acco
67. ion through the navigation key the phone connects the headset when the phone has an incoming call it will ring from the headset 5 8 Power Light 1 Press Menu gt Features gt Enter gt Power Light gt Enter 2 Enable this function through the navigation key 5 9 Hide DTMF 1 Press Menu gt Features gt Enter gt Hide DTMF gt Enter 2 Through the navigation key to choose Disabled All Delay Last Show When you set up a call with others and need to input the DTMF the DTMF will show as you have set 5 10 Ban Outgoing 1 Press Menu gt Features gt Ban Outgoing gt Enter 2 Enable this function you can not call any number 5 11 Password Dial 1 Press Menu gt Features gt Enter gt Password Dial gt Enter 2 Enable this function you can also set Prefix and Length For example you want call out 1234567 and you set Password Dial Prefix 123 and Password Length 3 then enter the dial interface and input 1234567 and then the screen will show 123 7 5 12 Pre Dial 1 Press Menu gt Features gt Pre Dial gt Enter 2 Enable this function you will realize Pre Dial function 5 13 Action URL amp Active URI 1 Action URL achieve results com from a functional understanding that end a phone Action produce a URL Action which means the side of the phone receieves incoming Incoming call outgoing calls Outgoing call turn DND open DND hang up the phone On hook etc To set the phone web
68. is number by line1 line Intercom function you can press this key in standby to automatically answer the call and make each other DSS Key 1 MemoryKey _ 14111 sipi 1 Intercom User can be configured in accordance with push to talk function the way 4116 was the other number Then press the standby button and make automatically answer the call 4116 key can be configured through the following events For example DSS Key 1 Key Event 55 DND 8 3 5 2EXT KEY EXT KEY nas the same usage with the Function key In port connects the phone Out port connects the next one if there is only you don t need for power supply if there are more than one you need supply 5V power for the first one and use 45 direct connector FUNCTION KEY emer SOFTKEY Expansion Module Selection Expansion Module 1 Fi None F2 None smi Noe ew F4 None px F5 None pe 1 Fe None r Tel 3 F8 None set F9 SFR F10 None sii F11 None SIP1 F12 None se F13 wn 14 None Fis jJ F16 None F17 None Fis F19 None F20 None F21 F22 F23 None F24 Nne F25 F26 None L Apply Connected None JE None JE None l
69. ks off the phone will dial to B Notice It needs a external software what supports click to dial 4 5 Call back The function allows you dial out the last phone call which you received 4 6 Auto answer When there is an incoming call after no answer time the phone will answer the call automatically 4 7 Hotline You can set hotline number for every sip and then enter the dialer interface and after Warm Line Time the phone will call out the hotline number automatically 4 8 Application 4 8 1 SMS 1 Press Menu gt Applications gt Enter gt SMS gt Enter 2 Use the navigation keys to highlight the options You can read the message in the Inbox Outbox 3 After view the new message you can press Reply to reply the message and use the 2aB softkey to change the Input Method when enter the reply message press OK then use the navigation keys to select the line from which you want to send then Send 4 If you want to write a message you can press New and enter message Use the 2aB softkey to change the Input Method When you input the message you want to send press OK then use the navigation keys to select the line from which you want to send then Send 5 If you want to delete the message after view the message press Del then you have three options to choose Yes All No 4 8 2 Memo You can add some memos to record some important things to remind you Press Menu gt Application gt Memo gt Enter gt Add
70. lick Delete to delete the selected IP segment Web Filter setting Select it or not to enable or disable Web Filter Click Apply to make it effective Notice Do not set your visiting IP outside the Web filter range otherwise you cannot logon through the web 8 3 7 2 FIREWALL Firewall Type Enable Input Rules Enable Output Rules E Apply Firewall Input Rule Table Index Deny Permit Protocol Src Address Src Mask Dest Address Dest Mask Range Port Firewall Output Rule Table Index Deny Permit Protocol Src Address Src Mask Dest Address Dest Mask Range Port Firewall Settings Input Output Input Src Address Deny Permit Deny Dest Address Protocol UDP Src Mask et ada Port Range more than Iz Dest Mask Rule Delete Option Input Output Input Index Be Deleted Firewall Configuration In this web interface you can set up firewall to prevent unauthorized Internet users from accessing private networks connected to the Internet input rule or prevent unauthorized private network devices from accessing the Internet output rule Firewall supports two types of rules input access rule and output access rule Each type supports at most 10 items Through this web page you could set up and enable disable firewall with input output rules System could prevent unauthorized access or access other networks set in rules for security Firewall is also called access lis
71. line calls Configuration Application examples 0 0 0 0 4569 LAX no suff 2T 0 0 0 0 5060 SIP de ni att The handset off hook exhale if SIP1 registration is successful the default is SIP1 If the dial 21111 then exhaled directly through SIP1 and the called number is 21111 If the phone off hook exhale if SIP1 registration is successful the default is SIP1 If dialing 32222 directly and through SIP1 outgoing called number is 32222 To make the configuration take effect dialpeer function Only when the handset off hook exhaled choose SIP2 and dials 21111 the corresponding rule is matched by SIP2 exhaled and the called number is 1111 Only when the handset off hook exhaled Select IAX2 and dials 32222 the corresponding rule is matched by IAX2 outgoing and called number is 2222 Examples of different alias application Set by web explanation example You need set phone number Destination Alias and Delete Length Phone number is XXXT Destination is If you dial UE RE 255 255 255 255 a eci aes 93333 the Alias optional del 0 0 0 2 and Alias is del ree eat SIP2 server will Delete Length optional 1 This means any phone receive 3333 No that starts with your set phone number will be sent via SIP2 line after the first several digits of your dialed phone number are deleted according to delete length Phone Number Destination optional Port o
72. ll automatically ignore all incoming multicast RTP stream Web page is set as follows MCAST Settings Priority 1 x Enable Page Priority Index Priority Name Host port 1 ss 239 1 1 1 1366 2 ee 239 1 1 1 1367 Now multicast ss has higher priority than multicast ee the highest priority is for normal calls Notice When a multicast session begins multicast sender and receiver will beep 8 3 5FUNCTION KEY 8 3 5 1 FUNCTION KEY Screen Configuration Contrast 5 1 9 Enable Backlight Line Key Settings Line Key Type Value Line Subtype Pickup Number tine E SiPi e None Line Key Line z 1 2 m None OE tine zl 5 5 None ZOEN Une spa None s Apply Function Key Settings Key Type Value Line Subtype Pickup Number DSS Key 1 Key Event 4 51 1 El Release x DSS Key 2 Key Event 4 SIP1 x MWI DSS Key 3 Key Event 4 SIP1 Headset z DSS Key 4 None E sIP1 None DSS Key 5 None E SIP1 None 055 6 E sipi_ None DSS Key 7 None 9 siP1 None DSS Key 8 None Iz siP1 None Programmable Key Settings Key Desktop Dialer Calling Desktop Long Pressed Up History x Prev Line z Prev Call Status x Down Status e
73. n on WAN and LAN port including the connect mode of WAN port Network Static DHCP PPPoE MAC address the IP address of WAN port and LAN port ON or OFF of DHCP mode of LAN port and bridge mod Accounts Shows the phone numbers provided by the SIP LINE 1 2 servers and IAX2 The last line shows the version number 8 3 12 WIZARD STATUS WIZARD CALL LOG LANGUAGE WAN Connection Mode Static IP DHCP Next Wizard Please select the proper network mode according to the network condition C58 C58P provide three different network settings Static If your ISP server provides you the static IP address please select this mode and then finish Static Mode setting If you don t know about parameters of Static Mode setting please ask your ISP for them DHCP In this mode you will get the information from the DHCP server automatically need not to input this information artificially PPPoE In this mode you must input your ADSL account and password You can also refer 02 2 1 Network setting to speed setting your network Choose Static IP MODE click NEXT can configure the network and SIP default SIP1 simply also can browse too Click BACK can return to the last page Static IP Settings IP Address Subnet Mask IP Gateway DNS Domain Primary DNS Secondary DNS 192 168 1 179 255 255 255 0 192 168 1 1 202 96 134 133 202 96 128 68
74. ne will send UDP message to server to keep alive every NAT Keep Alive Period s Keep Alive Interval Set examining interval of the server default is 60 seconds User Agent Set the user agent if have the default is VoIP Phone 1 0 DTMF Type Select DTMF sending mode there are three modes DTMF_RELAY DTMF_RFC2833 DTMF_SIP_INFO Different VoIP Service providers may provide different modes DTMF SIP INFO There are two options send 10 11 and send Mode Local Port Set sip port of each line Ring Type Set ring type of each line Enable Via Rport Enable Disable system to support RFC3581 Via rport is special way to realize SIP NAT Enable PRACK Enable or disable SIP PRACK function suggest use the default config Enable Long Contact Set more parameters in contact field connection with SEM server Convert URI Convert to 23 when send the URI Dial Without Set call out by proxy without registration Registered Ban Anonymous Call Set to ban Anonymous Call Enable DNS SRV Support DNS looking up with _sip udp mode Server Type Select the special type of server which is encrypted or has some unique requirements or call flows RFC Protocol Edition Select SIP protocol version to adapt for the SIP server which uses the same version as you select For example if the server is CISCO5300 you need to change to RFC2543 else
75. ness classmate pe Add Clear Import Contact List Select File Export Contact List Browse xml vcf csv Update Export XML Group Option Group friend Name friend Ring Type Default z Add Modify Delete Delete Blacklist Settings Blacklist Item E Delete Delete All Type Number Value Add Line Auto Field name explanation Phonebook Table Name Shows the name corresponding to the phone number Shows the detail of current phonebook Notice the maximum capability of the phonebook is 500 items you can select many contact to add to group and add to blacklist and delete many contact and delete all contacts Add Contact List Name Specify the name corresponding to the phone number Office Number Specify the office number Mobile Number Specify the mobile number Other Number Specify the other number Ring Type Specify the ring type for the phone number Line Specify the sip line for the each number Group setting Select the group from the unselected group to selected list for the contact you can select many groups for the contact Notice the add button for adding a new contact the modify button for modifying the added contact the clear all button for clear all input information of the contact Group Option
76. nt settings Proxy Server Port Set your Proxy SIP server port Proxy User Input your Proxy SIP server account Proxy Password Input your Proxy SIP server password Set the sip domain if needed otherwise this VoIP Domain Realm phone will use the Register server address as sip domain automatically Usually it is same with registered server and proxy server IP address Backup Server Address Input the Backup Server Address if the primary server is unavailable then the phone will enable the Backup Server Address Backup Server Port Specify the Backup Server Port Enable Registration Start to register or not by selecting it or not Codecs Settings Disable Use the navigation keys to highlight the desired Codecs Enable one in the Enable Disable Codecs list and press Codecs the desired to move to the other list Advanced SIP Setting Forward Type Select call forward mode the default is Disabled Off Close down calling forward Busy If the phone is busy incoming calls will be forwarded to the appointed phone No answer If there is no answer incoming calls will be forwarded to the appointed phone after a specific Always Incoming calls will be forwarded to the appoint phone immediately The phone will prompt the incoming while doing forward Forward Number Specify the number you want to forward No Answer Forward Wait Time S
77. ntication Ans With a Single Codec Auto TCP Enable Strict Proxy Enable GRUU Enable Displayname Quote Enable user phone Click To Talk Enable Group second s SIP Config 0 0 9 second s 60 second s 0 second s Local 3600 second s 60 second s COMMON RFC3261 5060 None Field name explanation SIP Line Choose line to set info about SIP there are 4 lines to choose You can switch by Load button Basic Settings Status Shows if the phone has been registered the SIP server or not or so show Unapplied Server Address Input your SIP server address Server Port Set your SIP server port Authentication User Input your SIP register account name Authentication Input your SIP register password Password SIP User Input the phone number assigned by your VoIP service provider Phone will not register if there is no phone number configured Display Name Set the display name Proxy Server Address Set proxy server IP address Usually Register SIP Server configuration is the same as Proxy SIP Server But if your VoIP service provider gives different configurations between Register SIP Server and Proxy SIP Server you need make differe
78. on is sent out from port 1 then transmitted to port 2 3and 4 In chart 2 red and blue indicate two different VLANs in the switch and port 1 and port 2 belong to red VLAN port 3 and port 4 belong to blue VLAN If a broadcast frame is sent out from port 1 switch will transmit it to port 2 the other port in the red VLAN and not transmit it to port3 and port 4 in blue VLAN By this means VLAN divide the broadcast domain via restricting the range of broadcast frame transition Note chart 2 use red and blue to identify the different VLAN but in practice VLAN uses different VLAN IDs to identify ENTER QoS amp VLAN SERVICE PORT DHCP SERVICE TIME amp DATE li Link Layer Discovery Protocol LLDP Settings Enable LLDP Packet Interval 1 3600 60 second s Enable Learning Function Quality of Service QoS Settings Enable DSCP SIP DSCP o 0 63 Audio RTP DSCP b 0 63 WAN Port VLAN Settings Enable WAN Port VLAN 7 WAN Port VLAN ID o 0 4095 SIP 802 1P Priority 0 0 7 Audio 802 1P Priority o 0 7 LAN Port VLAN Settings LAN Port VLAN Mode Follow WAN LAN Port VLAN ID o 0 4095 QoS amp VLAN Configuration Field name explanation Link Layer Discovery Protocol LLDP Settings Enabel LLDP Enable LLDP by selecting it Enable Learning After enabling LLDP Learn telephone can Funci n automatically learn the data of DSCP 802 1p VLAN ID from the swi
79. page you can config SIP STUN STUN STUN server the phone in private network could know the type of NAT and the NAT mapping IP and port of SIP The phone might register itself to SIP server with global IP and port to realize the device both calling and being called in private network Stun server tell ant to receive data from 5060 bort Gateway NAT STUN Server Simple Traversal of UDP through NATs STUN Settings STUN NAT Traversal FALSE Server Address m Server Port 8478 Binding Period 50 second s SIP Waiting Time 800 millisecond s Local SIP Port 5060 Apply SIP Line Using STUN SIP 1 Use STUN Apply Field name explanation Simple Traversal of UDP through NATs STUN Settings STUN NAT Traversal Shows STUN NAT Transverse estimation true means STUN can penetrate NAT while False means not Server Address Set your SIP STUN Server IP address Server Port Set your SIP STUN Server Port Blinding Period s Set STUN blinding period s If NAT server finds that a NAT mapping is idle after time out it will release the mapping and the system need send a STUN packet to keep the mapping effective and alive SIP Waiting Time Specify the sip wait stun time you can input the time depended on your network condition Local SIP Port Configuration the local SIP Port the default value is5060 this port immediate
80. pecify the No Answer Forward Delay Time if the Forward Type is No answer incoming calls will be forwarded after the no answer forward wait time Enable Hot Line Specify Hot Line by selecting it Hot Line Number Specify Hot Line Number the phone dial the hot line number automatically at hands free mode or handset mode after warm line time Warm Line Wait Time Specify the Warm Line Time Transfer Timeout For the phone supports the transfer of certain special features server set interval time between sending bye and hanging up after the phone transfers a call BLF Server Ordinary BLF application is that the phone send subscription package to the registered server if your server does not support subscription package please input the BLF server so that it can separate register server and BLF server SIP Encryption Enable Disable SIP Encryption SIP Encryption Key Set the key for sip encryption RTP Encryption Enable Disable RTP encryption RTP Encryption Key Set the key for RTP encryption Enable Auto Answer Enable Auto Answer by selecting it Auto Answer Timeout Specify Auto Answer Time the phone auto answers the incoming call after Auto Answer Time Enable Session Timer Set Enable Disable Session Timer whether support RFC4028 It will refresh the SIP sessions Session Timeout Set the session timeout Subsc
81. phone may not cancel call normally System uses RFC3261 as default Transport Protocol Set transport protocols TCP or UDP Anonymous call Edition Set Anonymous call out safely Support RFC3323and RFC3325 Keep Authentication Enable Disable Keep Authentication System will take the last authentication field which is passed the authentication by server to the request packet It will decrease the server s repeat authorization work if it is enable Answer With A Enable Disable the function when call is Single Codec incoming phone replies SIP message with just one codec which phone supports Auto TCP Set to use automatically TCP protocol to guarantee usability of transport as message is above 1300 byte Enable Strict Proxy Support the special SIP server when phone receives the packets sent from server phone will use the source IP address not the address in via field Enable GRUU Set to support GRUU Enable Display name Quote Set to make quotation mark to display name as the phone sends out signal in order to be compatible with server Enable user phone Enable user phone by selecting it it is contained in the invite sip message in order to be compatible with server Enable Missed Call Log Enable the missed call log by it the phone will save the missed call log into the call history record and display the missed calls on the idle screen or won
82. pplied or idle Table 4 Programmable key LEDs for MWI LED Status Description Blinking green There are new voice mails Off There is no new voice mail Table 5 Power Indication LED LED Status Description Steady red Power on Fast Blinking red There is an incoming call Off Power off 2 Initial connecting and Settings 2 1 Connect the power and network 2 1 1 Connect to network Please make sure your environment already have broadband internet access capability during this step 1 Broadband Router ae Internet gt QS Connect one end of the network cable to the ACOM214 s_ WAN port the other end is connected to your broadband router s LAN port so that the completion of the network hardware connections In most cases you must configure your network settings to DHCP mode The details setting mode please referto 2 2 1 Network Settings ADSL Cable Broadband Modem Router 2 No broadband router ee Internet SD Connect one end of the network cable to the ACOM214 s WAN port the other end is conneted to your broadband modem s LAN port so that the completion of the network hardware connections In most cases if you are using a TV cable broadband you must configure your network settings to DHCP mode if you are using ADSL you must set your ACOM214 to PPPOE mode The details setting mode please referto 2 2 1 Network Settings ADSL Cable Modem 3 Worked as a broa
83. presents go sipx line For compatibility with old code 0 0 0 0 means go sip1 line 255 255 255 255 indicates go sip2 line Configuration examples are as follows 2T 255 255 255 255 5060 SIP del no suffix 1 3T 0 0 0 0 4569 IAX2 del no suffix 1 If the phone dial 21111 the fact is through SIP2 and called number is 1111 If the phone dial 32222 the fact is through IAX2 and called number is 2222 2 Enable on line query capabilities Enable on line query function on the premise that The phone must be multi line products you can choose when dialing protocol and line So that each end of the dial and also selected protocol and line Dialpeer table in the query the first comparison dialing protocol is selected in the table and dialpeer agreement if the same continue down the match otherwise check the next one Step match line information comparing the selected dial up line is a line in the table and dialpeer is the same if the same continue down the match otherwise the next query The third step is for a prefix or exact match Mode to sip it means that this rule is only used for sip protocol calls iax2 it means that this rule is only used iax2 protocol calls Destination indicates the destination address 0 0 0 1 Indicates that the rule only calls for sip1 online 0 0 0 2 Indicates that the rule only calls for sip2 online 0 0 0 x Indicates that the rule only calls for sipX online 0 0 0 0 Indicates that the rules used in all on
84. ptional Alias optional all 33334444 This setting will realize speed dial function after you dialing the numeric When you dial 2 the SIP1 server will Call Mode SIP v Suffix optional Delete Length optional agn 1 key 2 the number receive after all will be sent out 33334444 The phone will automatically send out When you dial alias number adding 8309 the SIP1 Destination optional Port optional Alias optional add 0755 your dialed number if server will Call Mode SIP v Suffix optional elete Length optional your dialed number receive starts with your set 07558309 phone number You need set Phone When you dial Number Alias and u Phone Number 010T 0106228 the crus Delete Length Phone ort aptiona Alias optional rep 0086 SIP1 server will Call Mode SIP suffix ontional number is XXXT and Delete Length optional 3 Alias is rep If your dialed phone receive 86106228 number starts with your set phone number the first digits same as your set phone number will be replaced by the alias number specified and New phone number will be send out Phone Number Destination optional Port optional Alias optional Call Mode Suffix optional Delete Length optional SIP v 0011 If your dialed phone number starts with your set phone number The phone will send out your di
85. ribe for MWI Enable the Subscribe for MWI by selecting it the phone will send subscribe message for MWI to the SIP Server MWI Number Specify the MWI Number Please contact your system administrator for the connecting code Different systems have different codes Subscribe Period s Overtime of resending subscribe packet Suggest using the default configuration Conference Type Specify the Conference Type if you select the local you needn t input the conference number Conference Number Specify the network conference number please contact your system administrator for the network conference number Registration Expire s Set expire time of SIP server register default is 60 seconds If the register time of the server requested is longer or shorter than the expired time set the phone will change automatically the time into the time recommended by the server and register again Enable Service Code If you want to realize the following function by the server please enter the On Code and Off Code option then when you choose to enable disable following function on your IP phone it will send message to the server and the server will turn on off the function immediately DND On Code Set the DND On Code When you press the DND hot key the phone will send a message to the server and the server will turn on the DND function Then any calls to the extension will be rejec
86. rt 4569 Voice Mail Number 0 Voice Mail Text mail Echo Test Number 1 Echo Test Text echo Refresh Time 60 second s Enable Registration Enable G 729AB E IAX2 Config Field name explanation Status Shows if the phone has been registered the IAX2 server or not Server Address Input your IAX2 server address Server Port Set your IAX2 server port the default is 4569 Account Input your IAX2 register account name Password Input your IAX2 register password Phone Number Input your assigned phone number usually it is same you re your IAX2 account name Local Port Set your local sport the default is 4569 Voice Mail Specify the voice mail s number Number Voice Mail Text Specify the voice mail s name Echo Test Number Set echo test number If IAX2 server supports echo test and echo test number is non numeric system could set an echo test number to replace the echo test text So user can dial the numeric number to test echo voice test This function is provided with server to make endpoint to test whether endpoint could talk through server normally Echo Test Text Specify echo test text s name Refresh Time Set expire time of IAX2 server register you can set it between 60 and 3600 seconds Enable Start to register the IAX2 server or not by selecting it Registration or not Enable G 729AB Enable or disable code G 729 by selecting it or not 8 3 3 3 Stun In this web
87. s and port User will has different access level with different username and password Default user with general level Username guest Password guest Default user with root level Username admin Password admin The default password of phone screen menu is 123 8 2 Setting via web browser When this phone and PC are connected to network enter the IP address of the wan port in this phone as the URL e g http xxx xxx xxx xxx or http XXX XXX XXX XXX XXXX If you do not know the IP address you can look it up on the phone s display by pressing Status button The login page is as below picture User Password Language English v Logon After you configure the IP phone you need click save button in config under Maintenance in the left catalog to save your configuration Otherwise the phone will lose your modification after power off and on 8 3 Configuration via WEB 8 3 1 BASIC 8 3 1 1 STATUS STATUS WIZARD CALL LOG LANGUAGE Network WAN LAN Connection Mode DHCP IP Address 192 168 10 23 MAC Address 00 a8 59 cc b2 fc DHCP Service Enabled IP Address 192 168 2 5 Bridge Mode Disabled IP Gateway 192 168 2 1 Accounts SIP Line 1 5060 Unapplied SIP Line 2 5060 Unapplied SIP Line 3 5060 Unapplied SIP Line 4 5060 Unapplied SIP Line 5 5060 Unapplied SIP Line 6 5060 Unapplied IAX2 4569 Unapplied Status Field name Explanation Shows the configuration informatio
88. t is a simple implementation of a Cisco like access list firewall It supports two access lists one for filtering input packets and the other for filtering output packets Each kind of list could be added 10 items We will give you an instance for your reference Field name explanation Enable Input Rules Select it to Enable Input Rules Enable Output Select it to Enable Output Rules Rules Input Output Specify current adding rule by selecting input rule or output rule Deny Permit Specify current adding rule by selecting Deny rule or Permit rule Protocol Filter protocol type You can select TCP UDP ICMP or IP Port Range Set the filter Port range Src Address Set source address It can be single IP address network address complete address 0 0 0 0 or network address similar to 0 Des Address Set the destination address It can be IP address network address complete address 0 0 0 0 or network address similar to Src Mask Set the source address mask For example 255 255 255 255 means just point to one host 255 255 255 0 means point to a network which network ID is C type Dest Mask Set the destination address mask For example 255 255 255 255 means just point to one host 255 255 255 0 means point to a network which network ID is C type Click the Add button if you want to add a new output rule Then enable out access and click th
89. t the information from the DHCP server automatically need not to input this information artificially PPPoE In this mode you must input your ADSL account and password You can also refer to 2 2 1 Network setting to speed setting your network Select it to use DHCP mode to get DNS address if Obtain DNS server 3 you don t select it you will use static DNS server automatically The default is selecting it WAN Settings Static IP DHCP PPPoE IP Address 192 168 1 179 Subnet Mask 255 255 255 0 IP Gateway 192 168 1 1 DNS Domain Primary DNS 202 96 134 133 Secondary DNS 202 96 128 68 Apply If you user static mode you need set it IP Address Input the IP address distributed to you Subnet Mask Input the Netmask distributed to you IP Gateway Input the Gateway address distributed to you Set DNS domain postfix When the domain which you input cannot be parsed phone will DNS Domain automatically add this domain to the end of the domain which you input before and parse it again Primary DNS Input your primary DNS server address Secondary DNS Input your standby DNS server address Static IP DHCP PPPoE Service Name ANY User juser123 Apply If you uses PPPOE mode you need to make the above setting Service Name It will be provided by ISP User Input your ADSL account Password Input your ADSL password Note
90. tch If the data is different from the data of the LLDP server telephone will change its own value as the value of the switch Synchronous with VLAN in switch Package Interval 1 3600 The time interval of sending LLDP Packet Quality of Service QOS Settings Enable DSCP Enable DSCP by selecting it SiP DSCP Specify the value of the SIP DSCP Audio RTP DSCP Specify the value of the Audio RTP DSCP WAN Port VLAN Settings Enable WAN Port VLAN Enable WAN Port VLAN by selecting it WAN Port VLAN ID Specify the value of the WAN Port VLAN ID the range of the value is 0 4095 SIP 802 1p Priority Specify the value of the sip 8021 p priority the range of the value is 0 7 Audio 802 1p Priority Specify the value of the audio 802 1p priority the range of the value is 0 7 LAN Port VLAN Settings Follow WAN Follow the WAN ID Disable Disable Port VALN LAN Port Vlan Enable Enable Port VLAN and specify the Port VLAN ID different from WAN ID Specify the value of the Port VLAN ID different LAN Port VLAN ID from WAN ID the range of the value is 0 4095 8 3 2 4 Service Port You can set the port of telnet HTTP RTP by this page SERVICE PORT DHCP SERVICE TIME amp DATE Service Port Settings t Web Server Type HTTP HTTP Port BO HTTPS Port 443 Telnet Port 23 RTP Port Range Start 10000 RTP Port Quantit
91. ted by the server automatically And the incoming call record will not be displayed in the Call History DND Off Code Set the DND Off Code When you press the DND hot key the phone will send a message to the server and the server will turn off the DND function Always CFwd On Code Set the Always CFwd On Code when you choose to enable the always forward function on your phone it will send message to the server and the server will turn on the function immediately When there are calls to the extension the server will always forward it to the set number automatically And the IP phone will not show the record in the call history anymore Always CFwd Off Code Set the Always CFwd Off Code when you choose to disable the always forward function on your phone it will send message to the server and the server will turn off the function immediately Busy CFwd On Code Set the Busy CFwd On Code when you choose to enable the busy forward function v on your phone it will send message to the server and the server will turn on the function immediately When there are calls to the extension the server will forward it to the set number automatically based the forward type And the IP phone will not show the record in the call history anymore Busy CFwd Off Code Set the Busy CFwd Off Code when you choose to disable the busy forward function on your phone it will send message to the server
92. ting message is twelve English characters and five Chinese characters 8 3 2 NETWORK 8 3 2 1 WAN i WAN QoS amp VLAN SERVICE PORT DHCP SERVICE TIME amp DATE WAN Status Active IP Address 192 168 2 5 Current Subnet Mask 259 299 299 0 Current IP Gateway 192 168 2 1 MAC Address 00 a8 59 cc b2 fc MAC Timestamp 20130426 WAN Settings Obtain DNS Server Automatically Enabled z Static IP DHCP PPPoE Apply 802 1X Settings User admin Password hahahaha BEND Enable 802 1X 7 WAN Status WAN Status Active IP Address 192 168 2 5 Current Subnet Mask 255 255 255 0 Current IP Gateway 192 168 2 1 MAC Address 00 a8 59 cc b2 fc MAC Timestamp 20130426 Active IP Address The current IP address of the phone Curren Subenet Mask MAC Address The current MAC address ofthe phone Current IP Gateway The current Gateway IP address MAC Timestamp Shows the time of getting MAC address The current Netmask address WAN Settings Obtain DNS Server Automatically Enabled Static IP DHCP PPPoE Apply Please select the proper network mode according to the network condition C58 C58P provide three different network settings Static If your ISP server provides you the static IP address please select this mode and then finish Static Mode setting If you don t know about parameters of Static Mode setting please ask your ISP for them DHCP In this mode you will ge
93. tiple Way Call If the phone in the hook handsfree headset mode you can press Hold key input delay The screen will display a One on behalf of two seconds For example if you enter 123 45 the 123 indicates that the calls is successful wait 2 seconds to send DTMF 45 and 123 45 shows the interval of 6 seconds and so on Note The function key must be configured as a HOLD 3 11 Multi line In this phone you can registe 2 SIP account numbers and the 2 accuonts can be used at the same time There are four Keys used as SIP line toleranted to make calls in SIP accounts It will blink when the account registed failed In order to convenience the enterprise the phone support multiple call answering call hold and multi line call The user can answer 10 incoming call phones at most you can choose any call through pressing the fluctuation navigation key in taiking and the other 9 calls will be in held You also can press the fluctuation navigation key to change the call and recover the talking then last call will be held automatic You also can define the six line keys as multi line keys then each line key will relate to a call and you can choose the talking through pressing the line keys and recover the talking and the light to the line key will bright all the time when in taking then the light of the call in held is sparking If user has 4 line calls and wants to invite the five party during the call they can press Softkey Conf or Soft
94. ty File Select Security File Delete SIP TLS Files HTTPS Files OpenVPN Files Security Field name explanation Update Security File Select Security File Select the security file you want to update then click Update button to update Delete Security File Select Security File Select the security file you want to delete then click Delete button to update SIP TLS File Show SIP TLS authentication certification file HTTPS File Show HTTPS authentication certification file Open VPN Files Show Open VPN File authentication certification file 8 3 8 LOGOUT Logout Click Logout button to logout the system Click Logout and you will exit web page If you want to enter it next time you need input user name and password again 9 Appendix B Specification Hardware Item ACOM214 P Adapter Input 100 240V Input Output Output 5V 1A port WAN 10 100Base RJ 45 1 PORT LAN 10 100Base RJ 45 1 PORT Power Idle 2 5W Active 2 8W Consumption LCD Size 128x48 75x30mm Operation 0 40 C Temperature Relative Humidity 10 65906 CPU Broadcom VoIP chipset SDRAM 16M Flash 4M Dimension L x W x 260x255x65mm H Weight 1 07kg 9 1 2 Voice features SIP supports 2 SIP servers Support SIP 2 0 RFC3261 and correlative RFCs Support AX2 Support multiple call queuing Support IAX2 line key to call Codec G 711A u G 723 1
95. unt then press Enter you can do some sip settings 7 2 Network Press Menu gt Enter gt Advanced settings and then input the password to enter the interface Then choose Network and press Enter you can do network settings you can refer to 2 2 1 Network settings 7 3 Security Press Menu gt Enter gt Advanced settings and then input the password to enter the interface Then choose Security you can configure Menu Password Key lock Password Key lock Status and whether to ban Outgoing 7 4 Maintenance Press Menu gt Enter gt Advanced settings and then input the password to enter the interface Then choose Maintenance and press Enter you can configure Auto Provision Backup and Upgrade 7 5 Factory Reset Press Menu gt Enter gt Advanced settings and then input the password to enter the interface Then choose Factory Reset and press Enter you can choose Yes or No 8 Web Configuration 8 1 Introduction of configuration 8 1 1 Ways to configure ACOM214 has three different ways to different users Use phone keypad Use web browser recommendatory way Use telnet with CLI command 8 1 2 Password Configuration There are two levels to access to phone root level and general level User with root level can browse and set all configuration parameters while user with general level can set all configuration parameters except SIP 1 2 or IAX2 s that some parameters cannot be changed such as server addres
96. update download system update file 2 Config export Upload the config to FTP TFTP server name and save it 3 Config fie import Download the config file to phone from FTP TFTP server The configuration will be effective after the phone is reset 4 Phone book export vcf Upload the phonebook file to FTP TFTP server name and save it 5 PhoneBook import vcf Download the phonebook file to phone from FTP TFTP server Protocol Select FTP TFTP server Update Logo File Select File Specify the url of the logo file Delete Logo File Select File Select the logo that you want to delete Logo File Logo File Show the logo file 8 3 6 5 ACCESS You can add or delete user account and change the authority of each user account in this web page u AUTO PROVISION SYSLOG CONFIG UPDATE ACCESS REBOOT LCD Menu Password Settings Menu Password Apply Keyboard Lock Settings PIN to Lock Keyboard Password lees Apply Enable Keyboard Lock User Settings User User Level admin Root guest General Add User User Es M Confirm User Level Root x User Management admin Delete Modify Access Configuration Field name explanation Keyboard Set the password for entering the setting menu of Password the phone by the phone s key board The password is digit User Settings User User Level admin Root guest G
97. ver URL SIP Line User Password 1 2 Auto 3 Auto 4 Auto JL LDAP Settings LDAP Loap Ed Display Title Version version Server Address Server Port B89 Lane Authentication None Line AUTO v Username Password Search Base Enable Calling Search Telephone ttelephoneNumber Mobile mobile Other home Display Name len You need to match a XML Phonebook address and you can directly access to the corresponding remote phonebook on the phone For example Set the Phonebook Name as Ejointech Server URL is tftp 192 168 1 3 admin phonebook index xml Or Set the Phonebook Name as ldap Server URL is Idap 192 168 1 3 dc winline dc com Remote Phonebook Setting Phonebook Name Custom the phonebook name displayed on the phone Server URL Specify the server url of the remote phonebook SIP Line Specify the sip line for the remote phonebook Authentication Specify the authentication mode for remote phonebook User password Input the authentication username and password 8 3 4 6 WEB DIAL FEATURE DIAL PLAN CONTACT REMOTE CONTA WEB DIAL memi Web Dial Settings Dial Number e Hangup Line Selection Ed You can make a call through the WEB DIAL enter the Dial Number then press Dial if you want to finish the talk press Hang up 8 3 4 7
98. y 20 Service Port Field name explanation Service Port Settings Web Server Type Specify Web Server Type Set web browser port the default is 80 port if you HTTP Port want to enhance system safety you d better change it into non 80 standard port 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 Before using the https you must download https authentication certification into the phone then set web browser port the default is 443 port if you HTTPS Port want to enhance system safety you d better change it into non 443 standard port You can access to the web in https after rebooting the phone Telnet Port Set Telnet Port the default is 23 RTP Port Range Start Set the RTP Start Port It is dynamic allocation RTP Port Quantity Set the maximum quantity of RTP Port the default is 200 Notice 1 You need save the configuration and reboot the phone after set this page 2 Please REBOOT the system if you modify the HTTP or telnet port number the new number should be greater than 1024 3 If you set 0 for the HTTP port it will disable HTTP service 8 3 2 5 DHCP SERVICE DHCP Client Table Leased IP Address Client MAC Address DHCP Lease Table Name Start IP End IP Leased Time Subnet Mask IP Gateway DNS an 192 168 10 24 192 168 10 53 1440 255 255 255 0 192 168 10 23 192 168 10 23 DHCP

Ejointech ACOM214P

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