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ACOM212 VoIP Phone User Manual

1. all 33334444 be sent via SIP2 line after the first several digits of your dialed phone number are deleted according to delete length This setting will realize speed dial function after you dialing the numeric key 2 the number after all will be sent out The phone will automatically send out alias number adding your dialed number if your dialed number starts with your set phone number You need set Phone Number Alias and Delete Length Phone number is XXXT and Alias is rep xxx If your dialed phone 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 If your dialed phone number starts with your set phone number The phone will send out your dialed phone number adding suffix number When you dial 2 the SIP1 server will receive 33334444 When you dial 8309 the SIP1 server will recelve 07558309 When you dial 0106228 the SIP1 server will recelve 86106228 When you dial 147 the SIP1 server will recelve 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 Audio Settings First Codec Third Codec Fifth Codec Onhook Time Handset Volume Speakerphone Volume Headset Volume G 729AB Payload Length G 722 Timestamps Enable VAD Field name
2. User Settings Liser admin quest Liser Level Root General This table shows the current user existed User User Level Set account user name Set user level Root user has the right to modify configuration General can only read Password Confirm Set the password 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 FIREWALL Web Filter Table Start IP Address End IP Address Option Web Filter Table Settings Start IP Address E End IP Address E Web Filter Setting Enable Web Filter El 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
3. 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 Type Set DTMF Payload 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 AUDIO FEATURE DIAL PLAN CONTACT REMOTE CONTA WEB DIAL Feature Settings DND Do Not Disturb Disabled Ban Outgoing A Enable Call Transfer Y Enable Call Waiting v Semi Attended Transfer 9 Enable 3 way Conference W Enable Auto Handdown Y Accept Any Call W Auto Handdown Time 3 second s Enable Call Completion al Enable Auto Redial FJ Enable Pre Dial Y Auto Redial Interval 10 1 180 second s Enable Silent Mode E Auto Redial Times 10 1 100 Hide DTMF Disabled Auto Headset Y Ring From Headset E Enable Intercom Y Enable Intercom Mute E Enable Intercom Tone vi Enable Intercom Barge Y P2P IP Prefix DND Return Code 480 Temporarily Not Available Turn Off Power Light Busy Return Code 486 Busy Here Emergency Call Number 110 Reject Return Code 603 Decline y Enable Password Dial E Active URI Limit IP Password Dial Prefix Push XML Server Password Length o 0 31 Enable Call Waiting
4. PnP Transport PnP Interval Phone Flash Server Address Config File Name Protocol Type Update Interval Update Mode TRO69 Settings Enable TRO69 ACS Server Type ACS Server URL ACS User ACS Password TRO69 Auto Login Inform Sending Period 8 3 6 2 SYSLOG 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 Specify the PnP Server Specify the PnP Server Specify the PnP Transfer protocol Specify the Interval time unit is hour Set FTP TFTP HTTP server IP address for auto update The address can be IP address or Domain name with subdirectory 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 Specify the Protocol type FTP TFTP or HTTP Specify update interval time unit is hour Different update modes 1 Disable means no update 2 Update after reboot means update after reboot 3 Update at time interval means periodic update Enable TRO69 by selecting it Specify the ACS Server Type Specify the ACS Server URL Specify ACS User Specify ACS Password Enable TRO69 Auto Login by selecting it Specify the inform Sending Period unit is second Syslog is a p
5. phone book call records and use this number for quick dialing press this button you can dial quickly 4 A mm Hands free Make the phone into hands free mode NA Indicate This light will flash when there is a missed call light Keys combination include functions such as History Directory DND Menu Del Redial Sen d Soft key 1 2 3 4 Quit Answer Divert Reject Hold Transfer Co nf Close and so on u 4 5 101 ioe J Inputting the phone number or DTMF a 1 4 Port for connecting Port Name Description Power swtich Input 5V AC 1A WAN 10 100M Connect it to Network LAN 10 100M Connect it to PC Headset Port type RJ 9 connector 1 5 Icon introduction Icon Description Am Call out UTES Call in Call hold Auto answer Call mute Contact ase DND Do not Disturb i a In hand free mode In hook mode SMS Missed call rl Call forward 1 6 LED Status introduction Table 1 Power Indication LED LED Status Description Steady red Power on Fast Blinking red There is an incoming call Off Power off Z 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 Connect one end of the network cable to the ACOM212 s WAN port the other end is connected to your broadband router s LAN port so that the compl
6. 4 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 5 Other functions of ACOM212 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 Dial Plan 1 Press Menu gt Features gt Enter gt Dial Plan gt Enter 2 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 3 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 5 4 Auto Redial 1 Press Menu gt Features gt Enter gt Auto Redial
7. 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 OoS amp VLAN SERVICE PORT DHCP SERVICE TIME amp DATE Simple Network Time Protocol SNTP Settings Enable SNTP Enable DHCP Time E Primary Server Secondary Server DO O O Timezone GMT 08 00 Bening Ch Resync Period second s 12 Hour Clock E Date Format ijJanMon le Daylight Saving Time Settings Enable FJ Offset minutes s Month October y Week 5 e 5 iw Day Sunday wl Hour Manual Time Settings Year AAA Month Day po Hour A Minute po 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 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
8. 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 38 59 0c 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 TAX 4569 Unapplied Status Field name Explanation Shows the configuration information 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 Shows the phone numbers provided by the SIP Accounts LINE 1 2 servers and IAX2 The last line shows the version number 8 3 1 2 WIZARD STATUS WIZARD CALL LOG LANGUAGE WAN Connection Mode Static IP DHCP a PPPoE Next Wizard Please select the proper network mode according to the network condition ACOM212 provide three different network settings O 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 O DHCP In this mode you will get the information from the DHCP server automatically need not to input this information artifi
9. ACOM212 can immediately start connecting with each other When you hear the beep beep long beep the other phone started ringing until the other party pick up the handset or use the speakerphone when you can start talking When the call is finished press the Speakerphone key to end the call 4 Using the Redial button If you try to call over the telephone you can press Redial key to call a recently dialed number one Note that you restart the phone the system will clear the call log dial Redial key at this time will be invalid 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 2 If you are on a call currently press the answer softkey During the conversation you can alternate between Headset 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 EHE 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
10. 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 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 dialed 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 90000000 911 gaT 991ix 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 seco
11. ID o 0 4095 Apply QoS VLAN Configuration explanation Enable LLDP by selecting it After enabling LLDP Learn telephone can automatically learn the data of DSCP 802 1p VLAN ID from the switch 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 The time interval of sending LLDP Packet Enable DSCP by selecting it Specify the value of the SIP DSCP Specify the value of the Audio RTP DSCP Enable WAN Port VLAN by selecting it VLAN WAN Port VLAN ID SIP 802 1p Priority Audio 802 1p Priority LAN Port VLAN Settings LAN Port Vlan LAN Port VLAN ID 8 3 2 4 Service Port Specify the value of the WAN Port VLAN ID the range of the value is 0 4095 Specify the value of the sip 8021 p priority the range of the value is 0 7 Specify the value of the audio 802 1p priority the range of the value is 0 7 Follow WAN Follow the WAN ID Disable Disable Port VALN Enable Enable Port VLAN and specify the Port VLAN ID different from WAN ID Specify the value of the Port VLAN ID different from WAN ID the range of the value is 0 4095 You can set the port of telnet HTTP RTP by this page Service Port Settings Gp Web Server Type HTTP Port HTTPS Port Telnet Port RTP Port Range Start RTP Port Quantity Field name Service Port Settings Web Server Type HTTP Port 200 Apply Servi
12. 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 AUDIO FEATURE DIAL PLAN CONTACT REMOTE CONTA WEB DIAL MCAST Web Dial Settings Dial Number Line Selection 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 MCAST AUDIO FEATURE DIAL PLAN CONTACT REMOTE CONTA WEB DIAL MCAST MCAST Settings Priority 1 Enable Page Priority T Index Priority Name Host port 1 2 3 4 5 6 7 8 9 10 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 8 239 1 1 1 1366 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
13. Thank you for your purchasing ACOM212 Thank you for your purchasing ACOM212 ACOM212 is a full feature telephone that provides voice communication over the same data network that 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 ACOM212 may cause damage if you do not use a power adapter with ACOM212 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 1 3 Keypad Key Key name Function Description a a o Navigation key assist users for operating more m MAGERAN convinient EE 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 REDIAL Redial Outgoing Call 3 You could also find the specify contacts in
14. Tone Enable Call History 7 Enable Multi Line Enable Default Line Y Enable Auto Switch Line Allow IP Call v Play Talking DTMF Tone Vv Play Dialing DTMF Tone Apply 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 Unmute Missed Call IP Changed Idle To Busy Busy To Idle Block Out Settings Field name Do Not Disturb Ban Outgoing Enable Call Transfer Semi Attend ed Transfer ES pe Apply Black Out Add l Delete FEATURE explanation Select DND the phone will reject any incoming call the callers will be reminded by busy but any outgoing call from the phone will work well If you select Ban Outgoing to enable it and you cannot dial out any number Enable Call Transfer by selecting it Enable Semi Attended Transfer by selecting it Enable Auto Redial Auto Redial interval Auto Redial Times Enable
15. end day Hour Setup start and end hours Minute Setup start and end minutes Manual Time Settings Manual Time Settings Year Month Day fF Hour Minute Apply 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 SIP1 wr Basic Settings gt gt Status Unapplied Domain Realm O O Server Address Doo O OE Proxy Server Address O O Serwer Port Proxy Server Port O O Authentication User DOO O O Proxy User DO O O Authentication Password A Proxy Password SIP User Backup Server Address O O Display Name Doo O O Backup Server Port Enable Registration El Server Name A Codecs Settings gt gt Disabled Codecs 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 Disabled w 0 120 second s second s 3600 second s SIP Opt
16. every NAT Keep Alive Period s then the server responses with 200 to keep alive If the type is UDP the phone will send UDP message to server Keep Alive Interval User Agent DTMF Type DTMF SIP INFO Mode Local Port Ring Type Enable Via Rport Enable PRACK Enable Long Contact Convert URI Dial Without Registered Ban Anonymous Call Enable DNS SRV Server Type RFC Protocol Edition Transport Protocol Anonymous Call Edition Keep Authentication Answer WithA to keep alive every NAT Keep Alive Period s Set examining interval of the server default is 60 seconds Set the user agent if have the default is VoIP Phone 1 0 Select DTMF sending mode there are three modes O DTMF_RELAY O DTMF_RFC2833 O DTMF_SIP_INFO Different VoIP Service providers may provide different modes There are two options send 10 11 and send H Set sip port of each line Set ring type of each line Enable Disable system to support RFC3581 Via rport is special way to realize SIP NAT Enable or disable SIP PRACK function suggest use the default config Set more parameters in contact field connection with SEM server Convert to 23 when send the URI Set call out by proxy without registration Set to ban Anonymous Call Support DNS looking up with _sip udp mode Select the special type of server which is encrypted or has some unique requirements or call flows Select SIP protocol version to adapt for the SI
17. key the current call will turn out blind 7 User defined you can customize digital map rules to make dialing more flexible Itis 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 MITO FEATURE DIAL PLAN REMOTE CONTA WEB DIAL Basic Settings Y Press 2 to Send Dial Fixed Length 11 to Send Y Send after 5 o second s 3 30 Y Press to Do Blind Transfer Blind Transfer on Onhook Attended Transfer on Onhook Press DSS Key to Do Blind Transfer Apply Dial Plan Table Plans le _ Delete gt a a 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
18. 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 a special tunnel Wall Switchboard WEB FILTER FIREWALL Virtual Private Network VPN Status IP Address 0 0 0 0 VPN Mode Enable VPN El L2TP OpenvPN Layer 2 Tunneling Protocol L2TP VPN Server Address VPN User E VPN Password a Apply 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 Update Delete Security File Select Security File J SIP TLS Files HTTPS Files OpenVPN Files Security Field name explanation Update Security File Select Security File Select the security file you wan
19. the action message 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 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 those dial modes as followings 1 End with dial your desired number and then press 2 Fixed Length the phone will intersect the number according to your specified length 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
20. 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 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
21. 11 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 online calls Configuration Application examples 0 0 0 0 4569 Lane del 2T 0 0 0 0 5060 SIF del no suffix 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
22. 4 34 GALT FORWARD oionn A O 14 29 CALE HOLD iiaia aa a a 14 310 CAL WAN Coi ia ie 15 Bf CALL TRANSFER cis 15 3 8 THREE WAY CONFERENCE CALL esseesseesseesscesseesseesseessecssecssecssecssesssesssesssessee 15 4 ADVANCED FUNCTION OF ACOM212 2 0 ccccccccscccccccccvces 16 AN CLICKTO DIAL sisi aia ias 16 AZ AUTO ANSWER a an 16 43 HOTLINE ai 16 AA APPLICATION ai 16 AAL SMS a E e 16 AA MeMO os 17 AAS VOICE MA a SEENE RS 17 5 OTHER FUNCTIONS OF ACOM2 12 2 000 000 cccccccssscssssssssseees 18 S L AUTO HANDDOWN cies least bededocsssesesseancbees 18 D2 DALPLAN oasis 18 D DIALES O AE 18 SE AVTO RED Docs 18 5 9 GALE COMPLETION ssessesscuscessscccucueseushvasiecucesodcvacasaveseueccseesteasnecccescdecssesascsusesctoss 18 5 6 POWER LIGHT oai irie ss acecasts Jeecesdecucaesseekeuaiesexeuusteccesesesseecasseeke 19 5 7 RUDE WD ME ii ii le 19 58 PASSWORD DIA PE DUO PERA 19 5 9 ACTION URL amp ACTIVE URI cccccsccccscccccscscccscsvcccccccccscsccccsvcccccccccccssscsees 19 SLO PUSHAME a 20 6 THE BASIC SETTINGS OF ACOM212 sossesssssssesesesssesesesosssosos 20 Gil KEYBOARD cisne 20 O2 SCREEN DETTIN CSS ara 20 6 3 RING SEUTINGS caian 20 GA VOICE VOLUME id ciencia totiaiia 21 GS LIME ADATE Sent ia el dile ieiaei 21 0 0 GREETING WORDS secsec erone eener E EEEE 21 6 7 LANGUAGE uta ea Iii 21 7 ADVANCED SETTINGS OF ACOM212 cccccccccccccccccc
23. ACOM212 VoIP Phone User Manual 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 light
24. AGE Language Language Selection English Greeting Words Greeting Words x 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 greeting message is twelve English characters and five Chinese characters 8 3 2 NETWORK 8 3 2 1 WAN WAN OoS amp VLAN SERVICE PORT 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 38 59 0c b2 fc MAC Timestamp 20130426 WAN Settings Obtain DNS Server Automatically Enabled r Static IP DHCP PPPoE 302 1X Settings User Password El A z 5 gt TT E Enable 802 1X 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 fe MAC Timestamp 20130426 Active IP Address The current IP address of the phone Curren Subenet The current Netmask address Mask MAC Address The current MAC address of the phone Current IP Gateway The current Gateway IP address MAC Timestamp Shows the time of getting MAC address WAN Settings Obtain DNS Server Automatically Enabled Static IP DHCP PPPoE Apply Please select the proper network mode according to the network condition ACOM212 provide three differ
25. Auto Provision Settings Current Config Version Common Config Version CPE Serial Number 2 0002 2 0002 00100400xH020010000000010e597052 User user m 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 TRO69 Settings gt gt Apply DHCP Option Settings gt gt DHCP Option 66 l DHCP Option Setting Custom DHCP Option 128 254 Plug and Play PnP Settings gt gt Enable PnP w PnP Server E PnP Port 5060 PnP Transport PnP Interval Phone Flash Settings gt gt hour s Server Address Config File Name EA Protocol Type FTP Update Interval hourts Update Made Fanvil 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 Field name Auto Update Setting Current Config Version Common Config Version CPE Serial Number User Password Config Encrypt Key Common Config Encrypt Key Save Autoprovision Information DHCP Option Setting DHCP Option Setting Custom DHCP Option Auto Provision explanation Show the current config file s version If the version of the configuration downloaded is higher
26. Busy Incoming calls are immediately forwarded when the phone is busy No Answer Incoming calls are forwarded when the phone is not answered after a specific period 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 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 2 Attended Transfer During talk press the key Transf then input the number that you want to transfer to and press Send A
27. Call Completion Enable Pre Dial Enable Call Waiting Enable 3 way Conference Enable Call Waiting Tone Accept Any Call Enable Auto Hand down Auto Hand down Time Ring From Headset Enable Intercom Enable Intercom Mute Enable Intercom Tone Enable Intercom Barge Enable Enable Auto Redial by selecting it then the phone reminds whether redial when the caller is busy or rejects Specify the Auto Redial interval Specify the Auto Redial interval Enable Call Completion by selecting it Disable this feature in standby interface next number will realize the number rules send out over the time Enable the feature jthen the number will not be send out over the time Enable Call Waiting by selecting it Then the phone reminds 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 3 way conference by selecting it Disdale this function you will not hear the tone beep when there have multiple incoming calls If select it the phone will accept the call even if the called number is not belong to the phone The phone will hang up and return to the idle automatically at hands free mode Specify Auto Hand down Time the phone will hang up and 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 Enable Ring From Hands
28. 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 suffis Del Length 156 192 168 1 119 5060 SIP no allas no suffis O When you want to dial a long distance call to Beijing you need dial an 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 Allas Suffix Del Length LT 0 0 0 0 5060 SIP rep 010 no suffis 1 To save the memory and avoid abundant input of user add the follow functions Dial Peer Table Number Destination Port Mode Alias Suffix Deleted Length L3XXXKKKRKXX 0 0 0 0 5060 SIP add 0 no suffix O 13 5 9 xxxxxxxx 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 a
29. First Codec Second Codec Third Codec Fourth Codec Fifth Codec Sixth codec Onhook Time Default Ring Type Handset Output Volume Speakerphone FEATURE DIAL PLAN CONTACT REMOTE CONTA WEB DIAL Second Codec G 711U y G 729AB y Fourth Codec None y None y Sixth Codec None f 200 millisecond s Tone Standard China y 5 1 9 Default Ring Type Type 1 5 1 9 Headset Ring Volume 5 1 9 5 1 9 Speakerphone Ring Volume I5 1 9 20ms G 723 1 Bit Rate 6 3kb s 160 20ms DTMF Payload Type 101 96 127 Enable MWI Tone v Apply AUDIO Configuration explanation The first preferential DSP codec G 711A u G 722 G 723 1 726 32 G 729AB None The second preferential DSP codec G 711A u G 722 G 723 1 726 32 G 729AB None The third preferential DSP codec G 711A u G 722 G 723 1 726 32 G 729AB None The forth preferential DSP codec G 711A u G 722 G 723 1 726 32 G 729AB None The fifth preferential DSP codec G 711A u G 722 G 723 1 726 32 G 729AB None The sixth preferential DSP codec G 711A u G 722 G 723 1 726 32 G 729AB None Specify the least reflection time of Hand down the default is 200ms Set up the ring by default Specify Output receiver Volume grade 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
30. Key RTP Encryption RTP Encryption Key Enable Auto Answer Auto Answer Timeout Enable Session Timer Session Timeout Subscribe for MWI 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 Specify the number you want to forward Specify 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 Specify Hot Line by selecting it Specify Hot Line Number the phone dial the hot line number automatically at hands free mode or handset mode after warm line time Specify the Warm Line Time 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 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 Enable Disable SIP Encryption Set the key for sip encryption Enable Disable RTP encryption Set the key for RTP encry
31. P server which uses the same version as you select For example if the server is CISC05300 you need to change to RFC2543 else phone may not cancel call normally System uses RFC3261 as default Set transport protocols TCP or UDP Set Anonymous call out safely Support RFC3323and RFC3325 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 Enable Disable the function when call is Single Codec Auto TCP Enable Strict Proxy Enable GRUU Enable Display name Quote Enable user phone Enable Missed Call Log Click to talk Enable BLF List Use VPN BLF List Number SIP Global Settings Strict Branch Enable Group Registration Failure Retry Time incoming phone replies SIP message with just one codec which phone supports Set to use automatically TCP protocol to guarantee usability of transport as message is above 1300 byte 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 Set to support GRUU Set to make quotation mark to display name as the phone sends out signal in order to be compatible with server Enable user phone by selecting it it is contained in the invite sip message in order to be compatible with server Enable the missed call log
32. SECURITY Application Layer Gateway ALG Settings IPSec ALG FTP ALG PPTP ALG Apply Network Address Translation NAT Table Inside IP Address Inside TCP Port Outside TCP Port Inside IP Address Inside UDP Part Outside UDP Port NAT Table Option Transfer Type TCP Outside Port O Inside IP Address O O Inside Port Add 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 card and other equipment physical consultations speed testing speed under bridge mode near to 100M in order to ensure the quality of voice and communications
33. Shows if the phone has been registered the SIP server or not or so show Unapplied Input your SIP server address Set your SIP server port Input your SIP register account name Input your SIP register password Input the phone number assigned by your VoIP service provider Phone will not register if there is no phone number configured Set the display name 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 different settings Set your Proxy SIP server port Input your Proxy SIP server account Input your Proxy SIP server password Set the sip domain if needed otherwise this VoIP phone will use the Register server address as sip domain automatically Usually it is same with registered server and proxy server IP address Input the Backup Server Address if the primary server is unavailable then the phone will enable the Backup Server Address Specify the Backup Server Port Start to register or not by selecting it or not Use the navigation keys to highlight the desired one in the Enable Disable Codecs list and press the desired to move to the other list Forward Type Forward Number No Answer Forward Wait Time Enable Hot Line Hot Line Number Warm Line Wait Time Transfer Timeout BLF Server SIP Encryption SIP Encryption
34. able and save the result amended automatically according to the IP address and Netmask You need reboot the phone and the DHCP server setting will take effect Select NAT or not Select Port Mirror or not it only works in bridge mode the function of the port mirror is that copy 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 Ifyou choose the bridge mode the LAN configuration will be disabled 8 3 2 3 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 owitchboard un A sE __ the Broadcast Frame transmit to every other port except the send port Broadcast Frame Chart 1 Lise VLAN sf After Switchboard received the Broadcast Frame only transmit it to other port which belong to same VLAN with send port Switchboard Broadcast Broadcast Frame Frame MS VLAN 2 SY E 5 gt Domain pr au Chart 2 In chart 1 there is a layer 2 that switches without setting VLAN Any broadcast
35. ber 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 or 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 DIAL PLAN CONTACT REMOTE CONTACT WEB DIAL MCAST Remote Phonebook Settings Index Phonebook Name Server URL SIP Line User Password 1 Auto 2 0 __ Auto 3 Auto se 4 Auto Apply LDAP Settings LDAP LDAP 1 Display Title Version Version 3 Server Address Server Port 389 Authentication None Line AUTO y Username Password Search Base Enable Calling Search E Telephone telephoneNumber Mobile mobile Other home Display Name cn Apply 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 fanvil 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
36. bes how to configure the number IP table to achieve the configuration of multiple accounts simultaneously ST 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 aT 0 0 0 0 4564 1452 del reo suffis 1 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 of the 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 represents 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 En cl cane aaa 5060 SIP del no suffis 1 3T 0 0 0 0 4569 lan del no suffis 1 If the phone dial 21111 the fact is through SIP2 and called number is 11
37. bove 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 L3XXXXXXXKK 13 5 9 xxxxXKKX 156 1T Add Dial Peer Phone Number Destination Optional Port Optional Alias Optional Call Mode Suffix Optional Deleted Length Optional Dial Peer Option L3xxxxXxXxXXxXX v Field name Phone number Destination Port Alias Destination Port Mode Alias Suffix Deleted Length 0 0 0 0 5060 SIP add 0 no suffix 0 0 0 0 0 5060 SIP add 0 no suffix 0 192 168 1 119 5060 SIP no alias no suffix 0 0 0 0 0 5060 SIP rep 010 no suffix il SIP Apply modiy explanation 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 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
38. 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 t save the missed call log into the call history record and display the missed calls on the idle screen Set click to Talk need practical software support 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 Phone use vpn ip to communicate Specify the BLF List Number 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 by selecting it then the phone enable the sip group backup function Notice the deployment will become effective in all sip lines Specify the registration failure retry time ifthe 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 DIAL PEER fF E fo E oF TAX Status Unapplied Server Address Server Port 1509 Account Password Phone Number Local Port 4569 Voice Mail Number Voice Mail Text mall Echo Test Number Echo Test Text echo Refresh Time second s Enable Registrat
39. ccccees 22 TLE ACCOUN Sonan aaa aa a 22 Tal NETWORK meia A AE 22 Doe SE OPET ari 22 TA MAINTENANCE sasesesvecevsscustuasseesncucsduatnedsedeeussauatenicessasesscuaduaseesddeceteiateeseessecesceus 22 Z FACTORY RESE anuna a a a a a a e a 22 8 WEB CONFIGURATION cassssicccscccsceccteiccssdececeketereeciecacsesesticeestecticecce 23 8 1 INTRODUCTION OF CONFIGURATION eesseosseossesseesssesseesseesseesseosseosseessecssesssessees 23 SLL WAYS CO COMPUTE sosssscccsivaceseresavvontsrateccueiassnesanetesexsanceeadusaloccivanconesceeduans 23 S12 Password CONTISGUFAION inicia iia iia ideas 23 8 2 SETTING VIA WEB BROWSER ccccccccsscccsccccsscccsccccccscccsccccsscccccccccecccceccccssccesccess 23 8 3 CONFIGURATION VIA WEB sinaloa 24 Sb DAAC oerna O A EE 24 932 INE VV ORR Wy soenan anion E OEE ERORO 29 ee VOIR oaoa N T T ANR 39 ALONE a E A 55 SS PUNCTION KEY osc terres ct cc ssitececoeacecoshtesciceccscseceadeedecdeeucessshoes ciceaescsececdeedaster 70 0 360 Maln tenie sise ea A E A ea Eiaa 71 5 3 SECURITY aran a TEE O TEE NA 79 IDE ice 6 9 APPENDIX eter ne rere er Ne orn E 7 9 1 1 SPECIFICATION HARDWARE nia iaa iaa 87 py Al PAA SCO 2e2 LIL i A A er nn ear ne 7 9 1 3 Network features cias 88 9 1 4 Maintenance and management cssssssscccccssssssssssssccccsccccssssssssccosees 89 E 9 2 DIGIT CHARACTER MAP TABLE esesecsesesesecsesesesecsoseseseososeseseoscseseseceosesesessosos 89 1 Introducing ACOM212 VOIP Phone 1 1
40. ce Port explanation Specify Web Server Type Set web browser port the default is 80 port if you want to enhance system safety you d better change it into non 80 standard port Example The IP address is 192 168 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 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 Set the RTP Start Port It is dynamic allocation Start RTP Port Quantity Set the maximum quantity of RTP Port the default is 200 HTTPS Port 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 MITO QoSSWLAN SERVICE PORT DHCP SERVICE DHCP Client Table Leased IP Address Client MAC Address DHCP Lease Table Mame Start IP End IP Leased Time Subnet Mask IP Gateway DNS lan 192 168 10 24 192 168 10 53 1440 233 233 293 0 192 168 10 23 192 168 10 23 DHCP Lease Table Settings Leased Table Name Start IP Address O End IP Addres
41. cess other networks set in rules for security Firewall is also called access list 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 Rules Input Output Deny Permit Protocol Port Range Src Address Des Address Src Mask Dest Mask Select it to Enable Output Rules Specify current adding rule by selecting input rule or output rule Specify current adding rule by selecting Deny rule or Permit rule Filter protocol type You can select TCP UDP ICMP or IP Set the filter Port range Set source address It can be single IP address network address complete address 0 0 0 0 or network address similar to 0 Set the destination address It can be IP address network address complete address 0 0 0 0 or network address similar to 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 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 rul
42. cially PPPoE In this mode you must input your ADSL account and password You can also refer to2 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 IP Address Subnet Mask IP Gateway DNS Domain Primary DNS Secondary DNS Quick SIP Settings Display Name Server Address server Port Authentication User Authentication Password SIP User Enable Registration Display Name Server Address Server Port Authentication User Authentication Password SIP User Enable Registration 192 168 1 179 192 168 1 1 202 96 134 133 202 96 128 68 lt lt Next Input the IP address distributed to you Input the Netmask distributed to you Input the Gateway address distributed to you Set DNS domain postfix When the domain which 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 Input your primary DNS server address Input your standby DNS server address Back l Next Set the display name Input your SIP server address Set your SIP server port Input your SIP register account name Input your SIP register password Input the phone number assigned by y
43. click Add to add this IP segment You can also click 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 WEB FILTER FIREWALL Firewall Type Enable Input Rules El Enable Output Rules El _ 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 y Src Address Deny Permit Deny Dest Address O O pedo Add Protocol upp Src Mask Ll Port Range more than e _ Dest Mask Rule Delete Option Input Output Input Index To Be Deleted e _ Delete 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 ac
44. dial to B Notice It needs a external software what supports click to dial 4 2 Auto answer When there is an incoming call after no answer time the phone will answer the call automatically 4 3 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 4 Application 4 4 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 4 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 There are some options to configure Mode Date Time text Ring When the configuration is completed press Save 4
45. e Then enable out access and click the 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 ye SUEI uot NAT Equipment Inner network E Private IP E 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 Inner Network area WEB FILTER FIREWALL NAT
46. eb Configuration 8 1 Introduction of configuration 8 1 1 Ways to configure ACOM212 has three different ways to different users O Use phone keypad Use web browser recommendatory way O 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 address and port User will has different access level with different username and password O Default user with general level Username guest Password guest O 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 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
47. ent network settings O 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 O 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 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 l l l you don t select it you will use static DNS server Aay The default is selecting it WAN Settings Static IP DHCP PPPoE IP Address Subnet Mask IP Gateway l192 168 1 1 DNS Domain Primary DNS 202 96 134 133 Secondary DNS 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 renee 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 E DHCP PPPoE P Service Name User Password Apply _ If you uses PPPOE mode you need to make t
48. ent 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 O Upgrade firmware through HTTP FTP or TFTP Telnet remote management upload download setting file O Support Syslog O Support Auto Provisioning upgrade firmware or configuration file E 9 2 Digit character map table Keypad Keypad i a a ZABCabc 4 y 8TUVtuv 3DEFdef 4 y IWXYZwxyz 4GHIghi 4 el Ba II 6MNOmno 4 SEND
49. et by selecting it the phone plays ring tone from handset Enable Intercom Mode by selecting it Enable mute mode during the intercom call If the incoming call is intercom call the phone plays the intercom tone Enable Intercom Barge by selecting it the phone auto answers the intercom call during a call If the current call is intercom call the phone will reject the second intercom call Enable Silent Mode by selecting it the phone light will red Silent Mode Turn Off Power Light Emergency Call Number Enable Password Dial Password Length DND Return Code Busy Return Code Reject Return Code Hide DTMF Push XML Server P2P IP Prefix blink to remind that there is a missed call instead of playing ring tone Enable Turn Off Power Light by selecting it Specify the Emergency Call Number Despite the keyboard is locked you can dial the emergency call number Enable Password Dial by selecting it When number entered is beginning with the password prefix the following N numbers After the password prefix will be hidden as N stand for the value which you enter in the Password Length field 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 Specify the Password length Specify DND Return code Specify Busy Return Code Specify Reject Return Code Specify the hide DTMF mode Specify the Push XML Server when
50. et the level of SIP log IAX2 Log Level Set the level of IAX2 log Enable Syslog Select it or not to enable or disable syslog Web Capture 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 8 3 6 3 CONFIG Click the end button to stop capturing the packet stream AUTO PROVISION SYSLOG UPDATE ACCESS Save Configuration Backup Configuration Clear Configuration Field name Save Configuration Backup Configuration Clear Configuration Click Save button to save the configuration files Save Save all network and VOIP settings Right Click here to Save as Config File txt Right Click here to Save as Config File xml Click Clear button to clear the configuration files Clear Config Setting Explanation You can save all changes of configurations Click the Save button all changes of configuration will be saved and be effective immediately Right clicks on Right click here and select Save 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 User can restore factory default configuration and reboot the phone If you login as Admin the phone will reset all configurations and restore factory default if you login as Guest the phone will reset all configurations exce
51. etion of the network hardware connections In most cases you must configure your network settings to DHCP mode The details setting mode please refer to 2 2 1 Network Settings pe Internet gt gt ADSL Cable Broadband Modem Router IP Phone 2 No broadband router Connect one end of the network cable to the ACOM212 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 ACOM212 to PPPOE mode The details setting mode please refer to 2 2 1 Network Settings are Internet xs ADSL Cable Modem IP Phone 3 Worked as a broadband router ACOM212 have broadband routing capability as long as the ACOM212 properly connected to the WAN port on the broadband modem and connect your computer or other Internet capable devices connected to the ACOM212 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 Mntemer gt a ADSL Cable Modem IP Phone lt S 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 socke
52. frame will be transmitted to the other ports except the send port For example a broadcast information 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 Qo0S amp VLAN SERVICE PORT DHCP SERVICE TIMES DATE Link Layer Discovery Protocol LLDP Settings Enable LLDP Enable Learning Function Quality of Service QoS Settings Enable DSCP Audio RTP DSCP WAN Port VLAN Settings Enable WAN Port VLAN SIP 802 1P Priority LAN Port VLAN Settings LAN Port VLAN Mode Field name Link Layer Discovery Protocol LLDP Settings Enabel LLDP Enable Learning Funcion Package Interval 1 3600 Quality of Service QOS Settings Enable DSCP SiP DSCP Audio RTP DSCP WAN Port VLAN Settings Enable WAN Port FJ Packet Interval 1 3600 second s E E SIP DSCP lo 0 63 od 0 63 E WAN Port VLAN ID o 0 4095 lo 0r7 Audio 802 1P Priority o 0 7 Follow WAN e LAN Port VLAN
53. fter 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 8 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 4 Advanced function of ACOM212 4 1 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 hooks off the phone will
54. 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 5 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 6 Power Light 1 Press Menu gt Features gt Enter gt Power Light gt Enter 2 Enable this function through the navigation key 5 7 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 8 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 t
55. he above setting Service Name It will be provided by ISP User Input your ADSL account Password Input your ADSL password Note 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 2 2 LAN LAN Settings g IP Address Subnet Mask DHCP Service NAT Port Mirror Enable Bridge Mode Field name LAN IP Address Subnet Mask DHCP Service NAT Port Mirror QoS amp VLAN SERVICE PORT DHCP SERVICE TIME amp DATE 192 168 10 23 293 299 233 0 Y 7 fl Only works in the bridge mode Apply LAN Config explanation Specify LAN static IP Specify LAN Netmask Select the DHCP server of LAN port or not After you modify the LAN IP address phone will amend and adjust the DHCP Lease T
56. he screen will show 123 7 5 9 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 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 PCIP then PC can push text SMS phonebook advertisement execute etc to phone to update the message or the phone makes an action 5 10 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 ACOM212 6 1 Keyboard 1 Press Menu gt Settings gt Enter gt Basic Settings gt Enter gt Keyboard gt Enter 2 There are fo
57. 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 You need set phone number Destination Alias and Delete Length laten T Phone number is XXXT If you dial Destination optional Port optional Destination is 93333 the Alias optional call Mode 255 255 255 255 SIP2 server will a 0 0 0 2 and Alias is del receive 3333 This means any phone No that starts with your set phone number will Phone Number Destination optional Port optional lias optional Call Mode SIP Suffix optional Delete Length optional Phone Number Destination optional Port optional Alias optional add 0755 Call Mode SIP y Suffiz optional Delete Lenoth optional Phone Number Destination optional Port optional Alias optional rep 0086 Call Mode SIP _ gt Suffixtoptional Delete Length optional 3 Phone Number Destination optional Port optional Alias optional Call Mode Suffiz optional Delete Length optional
58. ion i SEE A A A A A A RFC2833 Send 10 11 Default Enabled Codecs 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 Of Code No Ans CFwd Off Code Anonymous Off Code Keep Alive Interval Server Type RFC Protocol Edition Local Port Anonymous Call Edition Keep Authentication Ans With a Single Codec Auto TCP Enable Strict Proxy Enable GRUU Enable Displayname Quote Enable DNS SRV Enable user phone Enable Missed Call Lag Click To Talk SIP Global Settings gt gt Strict Branch Enable Group Registration Failure Retry Time second s Al 4 EE 0 9 second s second s oh second s Local 3600 second s ll Th second s COMMON RFC3261 None a pe pooooookgzf 3 EM Field name SIP Line SIP Config explanation Choose line to set info about SIP there are 4 lines to choose You can switch by Load button Basic Settings Status Server Address Server Port Authentication User Authentication Password SIP User Display Name Proxy Server Address Proxy Server Port Proxy User Proxy Password Domain Realm Backup Server Address Backup Server Port Enable Registration Codecs Settings Disable Codecs Enable Codecs Advanced SIP Setting
59. ion EJ 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 Set echo test number If IAX2 server supports echo Echo Test test and echo test number is non numeric system Number 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 page you can config SIP STUN STUN By 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
60. nds 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 AUDIO FEATURE DIAL PLAN CONTACT REMOTE CONTA WEB DIAL MCAST Phonebook Table Group All wm Hangup Index Name Office Number Mobile Number Other Number Ring Type Group E Page x Add to Blacklist Add Contact Name Ring Type Default Office Number Line Auto Mobile Number Line Auto Other Number Line Auto Group Setting Unselected Selected friend a a home J work F business classmate v v Add Import Contact List Select File xml vcf csv Export Contact List Group Option Group friend Name friend Ring Type Default Blacklist Settings Blacklist Item Type Number y Value Line Auto y Blacklist Contact 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 or a contact to add to group and add to blacklist and delete many or a contact and delete all contacts Add Con
61. nfo DTMF Relay RFC2833 Support 9 systems ringtones and three user defined ringtongs Soft keys programmable 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 Code synchronization via IP PBX IMS Support click to dial via web phone book Voice codec setting for each SIP line 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 cli
62. ning 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 ACOM212 VOIP PHONE ccccccsccsccsccccces 6 1 1 THANK YOU FOR YOUR PURCHASING ACOM2 12 cccccsccsccscccsccccccsccsccccccscceces 6 1 2 DELIVERY CONTENT scccrssdes ts scdsestcsencasustecavensesides ceicdavsessercausssecensesioecsaassedsnseoasccens 6 DS RCE VPA iii 7 4 PORT FOR CONNECTING sccsessssecsssncicasecesseces cesceonssvesacsesueceveususseces bacetoasscesatsseaseesuc 8 1 5 TON INTRODUCTION sroine E 8 1 6 LED STATUS INTRODUCTION iaa 9 2 INITIAL CONNECTING AND SETTINGS cccccccccsccsccccceees 10 2 1 CONNECT THE POWER AND NETWORK cccosscccssccccccccccccccsscccscccccccccccccccsscccseccess 10 ZAM iaa A oina a aaa aaa 10 2 1 2 Power adapter connection sscsssssseececcccsssseceeoccocsssccececoossssseeeoosossssseee 11 2 2 BASIGINITIALIZATION iii 11 22L Network SOTIMOS id 11 3 THE BASIC FUNCTION OF ACOM2122 cccccccccccccccccceees 13 Bek MAKINGA CAT incio 13 ILE CAMDEN id 13 PI Cal Methods aicr aeaaea a eaaa hevedtacidensel iaai 13 32 ANSWERINGA CAL sanan 14 DS DND da E AE EAE AEREAS 1
63. number to realize calling to what your prefix number is mapped The prefix number supports at most 30 digits 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 Set the Signal port the default is 5060 for SIP 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 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 descri
64. 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 will automatically ignore all incoming multicast RTP stream Web page is set as follows MCAST Settings Priority PL is Enable Page Priority El Index Priority Name Host part 1 ss 239 1 1 1 1366 2 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 SOFTKEY Softkey Settings Softkey Mode More ml Screen Call Dialer A Unselected Softkeys Selected Softkeys None Call Back CBack Next Line Next Out Pause Phonebook Dir Pickup Prev Line Prev Redial SOFTKEY You can configure different functions in different screens for every softkey 8 3 6 Maintenance 8 3 6 1 Auto Provision CONFIG UPDATE REBOOT AUTO PROVISION SYSLOG CONFIG ACCESS
65. our VOIP service provider Start to register or not by selecting it or not Connection Mode Static IP Static IP Address 192 168 1 179 IP Gateway 192 168 1 1 SIP Server Address Account Phone Number Registration Disabled Back _ 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 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 User Password Back Next PPPOE Server Scie name if AON Service providers are no special 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 1 3 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 LANGU
66. phone receives request 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 want to dial is 192 168 1 119 If you define P2P IP Prefix as 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 ns 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 L Active URI Limit IP Action URL Settings Action URL Settings Block Out Settings Block out Apply Block Out add m Specify the server IP that remote control phone for corresponding operation Specify the Action URL that Record the operation of phone send this corresponding information to server url http InternalServer FileName xml Internal Server is server IP Filename is name of xml that contains
67. pt 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 AUTO PROVISION SYSLOG CONFIG UPDATE ACCESS REBOOT Web Update Select File z txt xml au vcf csv wav Update TFTP FTP Update Server Address i User ey Password Apply File Name rs Type Application Update e Protocol FTP e Update Logo File Select File Browse Update Delete Logo File Select File screensaver txt Logo File screensaver Ext 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 Af
68. ption Enable Auto Answer by selecting it Specify Auto Answer Time the phone auto answers the incoming call after Auto Answer Time Set Enable Disable Session Timer whether support RFC4028 It will refresh the SIP sessions Set the session timeout Enable the Subscribe for MWI by selecting it the phone will send subscribe message for MWI to MWI Number Subscribe Period s Conference Type Conference Number Registration Expire s Enable Service Code DND On Code DND Off Code Always CFwd On Code Always CFwd Off Code the SIP Server Specify the MWI Number Please contact your system administrator for the connecting code Different systems have different codes Overtime of resending subscribe packet Suggest using the default configuration Specify the Conference Type if you select the local you needn t input the conference number Specify the network conference number please contact your system administrator for the network conference number 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 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
69. ress 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 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 gt Settings gt 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
70. rotocol 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 which 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 AUTO PROVISION SYSLOG CONFIG UPDATE ACCESS REBOOT Syslog Settings Server Address Server Port 314 MGR Log Level SIP Log Level IAX2 Log Level None el Enable Syslog E apply Web Capture Start 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 S
71. s O O Leased Time O minute s Subnet Mask A IP Gateway DNS Server Address i sy Add DHCP SERVICE Field name explanation DHCP Lease Table IP MAC mapping table If the LAN port of the 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 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 Y Apply Enable DNS Relay Select DNS Relay the default is enabled Click the 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 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
72. t 2 Plug power adaptor s DC output to the DC5V port of ACOM212 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 ACOM212 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 ACOM212 uses DHCP to 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 p
73. t 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 9 1 1 Specification Hardware ACOM212 P Adapter Input 100 240V Input Output Output 5V 1A 10 100Base T RJ 45 1 PORT 10 100Base T RJ 45 1 PORT Consumption 62 x 22mm Temperature Relative Humidity 10 65 Broadcom VoIP chipset SDRAM 16MB Dimension L x W x 155X 185X 130mm 9 1 2 Voice features SIP supports 2 SIP servers Support SIP 2 0 RFC3261 and correlative RFCs Support IAX2 Support multiple call queuing Support IAX2 line key to call Codec G 711A u G 723 1 G 729a b G 722 1 G 726 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 Support DTMF type SIP i
74. tact 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 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 num
75. ter reboot other modules of system still use previous setting and are not lost Type Protocol Update Logo File Select File Delete Logo File Select File Logo File Logo File 8 3 6 5 ACCESS Action type that system want to execute 1 Application update download system update file 2 Config file export Upload the config file 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 Select FTP TFTP server Specify the url of the logo file Select the logo that you want to delete Show the logo file You can add or delete user account and change the authority of each user account in this web page AUTO PROVISION SYSLOG ACCESS REBOOT LCD Menu Password Settings Menu Password Keyboard Lock Settings PIN to Lock Keyboard Password Enable Keyboard Lock User Settings User admin guest Add User User Fassword Confirm User Level User Management admin e ae E User Level Root General e Root fel Access Configuration Field name Keyboard explanation Set the password for entering the setting menu of Password the phone by the phone s key board The password is digit
76. ter 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 ACOM212 support three languages you can use the navigation Keys to choose The default two languages are English and Chinese 7 Advanced Settings of ACOM212 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 Account 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 W
77. 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 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 configuration in the server is different with the running configuration Show CPE Serial Number Specify FTP HTTP HTTPS server Username System will use anonymous if username keep blank Specify FTP HTTP HTTPS server Password Input the Encrypt Key if the configuration file is encrypted Input the Common Encrypt Key if the Common Configuration file is encrypted Save the username and password authentication message of http https ftp and input ID message in the phone until the url in the server changes 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 A valid Custom DHCP Option is from 128 to 254 The Custom DHCP Option must be in accordance with the one defined in the DHCP server Plug and Play Enable PnP PnP Server PnP Port
78. the server and the server will turn on off the function immediately 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 rejected by the server automatically And the incoming call record will not be displayed in the Call History 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 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 Set the Always CFwd Off Code when you choose to disable the always forward function on your Busy CFwd On Code Busy CFwd Off Code No Answer CFwd On Code No Answer CFwd Off Code Anonymous On Code Anonymous Off Code Keep Alive Type phone it will send message to the server and the server will turn off the function immediately 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
79. to SIP server with global IP and port to realize the device both calling and being called in private network Send request to Stun server ant to receive data from 5060 Private Network Gateway NAT STUN Server DIAL PEER Simple Traversal of UDP through NATs STUN Settings STUN NAT Traversal Server Address Server Port Binding Period SIP Waiting Time Local SIP Port SIP Line Using STUN SIP 1 A Use STUN Field name Simple Traversal of UDP through NATs STUN Settings STUN NAT Traversal Server Address Server Port Blinding Period s SIP Waiting Time Local SIP Port Sip Line Using STUN FALSE second s millisecond s Apply STUN explanation Shows STUN NAT Transverse estimation true means STUN can penetrate NAT while False means not Set your SIP STUN Server IP address Set your SIP STUN Server Port 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 Specify the sip wait stun time you can input the time depended on your network condition Configuration the local SIP Port the default value is5060 this port immediate effect modify SIP call will use the modified port communication SIP Line Using STUN SIP Use STUN Apply Choose line to set info about SIP There are 2 lines to choose Use STUN Enable
80. 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 ACOM212 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 M 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 y 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 1 Speed Dial In standby mode you simply enter your number to dial and press or press Redial to make a call 2 Hook dialing Pick up the handset and hear dial tone you can start dialing After entering the destination number press the key ACOM212 can immediately start connecting with each other When you hear the beep beep long beep the other phone started ringing until the other party pick up the handset or use the speakerphone time of the call is displayed on the screen you can start talking When the call is completed replace the handset hang up the call 3 Hands free Dialing Press the speakerphone key and hear a dial tone you can start dialing After entering the destination number press the key
81. ur 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 Settings 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 en
82. 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 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 and the server will turn off the function immediately 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 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 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 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 Specify the keep alive type if the type is option the phone will send option sip message to server

ACOM212 VoIP Phone User Manual

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