AT-620 User Manual
Contents
1. Default username and password is Administrator Username admin password admin User Username guest Username guest ATCOM TECHNOLOGY CO LIMITED AT 620 User Manual ATC O wi 1 Current state IP Phone EE ATCOM Current Status Network VOIP Advanced Dial peer Config Manage Update System Manage Running Status ml pacaan ooer frese a OOO y CO RR O Registered Unapplied Unregistered Version VOIP PHONE V1 7 61 48 Apr 9 2009 17 47 28 This page shows the IP phone working status The network part shows the connection status of WAN and LAN Phone Number part shows the phone number and register status for Linel Line2 and IAX2 2 Network 2 1 Wan Config There are 3 ways to connect to the internet DHCP Static and PPPoE please choose one according to your own situation A DHCP the IP phone will get IP address from DHCP server you do not have to fill in the date of IP address net mask etc just choose DHCP and submit Please refer to the below picture IP Phone een H ATCOM Current Status Network VOIP Advanced Dial peer Config Manage Update System Manage e WAN Config WAN Configuation e LAN Config Active IP 1921681116 Current Netmask 255 255 255 0 Current Gateway 192 168 1 1 MAC Address 00 0e 22 55 11 68 WAN Setting static Je Jee Parameters ATCOM TECHNOLOGY CO LIMITED AT 620 User Manual ATC O wi Active IP IP phone s address Current Net mask network2. ATC O wi protection is not discarding year nor usage life H Operating Requirement gt Operation temperature O to 40 C 32 to 104 F gt Storage temperature 30 to 65 C 22 to 149 F gt Humidity 10 to 90 no dew 10 Packing List gt AT 620 IP phone gt Power adaptor output 12v 500mA gt Manual CD 11 Installation Use Ethernet cable to connect AT 620 s LAN port and your computer Set computer s IP to the network 192 168 10 x or using dynamic obtain IP Open web browser and key in 192 168 10 1 Then user will see the logon page of AT 620 the default username and password is admin admin for administrator and guest guest for guest Set up page for VolP user only ATCOM TECHNOLOGY CO LIMITED AT 620 User Manual ATC O wi 2nd Feet installation instruction 1 Desktop position A Put the bottom side of the IP phone upside and press the plate with letter PUSH into the slot please refer the picture as below D Disassemble the feet Press the plate with word PUSH and pull the feet with the direction of arrow When the plate is pull out of the slot there will be a sound of pa you can take off the feet 2 On wall postion A Put the bottom side of the IP phone upside and push the plate with letter PUSH into the slot please refer the picture as below ATCOM TECHNOLOGY CO LIMITED AT 620 User Manual ATC O wi C Repeat Aand B Itis the picture
3. Hold User can hold the current call by pressing soft button Hold And by pressing soft button Hold again user can get back to the previous call In 3 way conference call mode user can also press this button to hold 3 way conference call and if you press it again user can go back to 3 way conference mode If hang up without exiting the status of hold The conversation will not be cancelled the line is still on hold 8 3 Way Conference Calls Assume B is AT 620 phone among user A B and C A calls B and talks with B through VolP 1 B can press soft button conf to hold the call with A 2 Then B inputs C s number 3 B presses Soft button dial to call to C 4 Cis on the call with B and A is on hold 5 B presses Soft button Spli button to make 3 way conference call 6 B presses soft button spli to end 3 way conference call and returns to the call with A 7 B presses soft button exit to end all the calls H Call History AT 620 supports 100 missed calls incoming calls and dialed calls record When the storage is full the latest call will update the history When the phone reboots or be out of power all the call history will be cleared gt Missed call 1 When the LCD screen displays number Missed call s press soft button Miss then the screen shows Missed Call 2 Press soft button OK the phone displays missed call numbers 3 Press navigation button to brow
4. Network VOIP Advanced Dial peer Config Manage Update System Manage Config Manage Save Configuration Press the Save button to save the configuration files 8 SE Save Backup Config Save all Network and vol settings Right Click here to Save as Config File txt Clear Configuration Press the Clear button to Clear the configuration files Clear Save Config you can Save all changes of configurations Click the Save button all changes of configuration will be saved and be effective immediately Backup Config Right clicks on Right click here and select Save Target As then you will save the config file in txt format Clear Config user can restore factory default configuration and reboot the phone If you login as Admin the phone will reset all configurations and restore factory default if you login as Guest the phone will reset all configurations except for VolP accounts SIP1 SIP2 and IAX2 and version number ATCOM TECHNOLOGY CO LIMITED AT 620 User Manual ATC O wi 7 Update 7 1 Web Update IP Phone Sege ATCOM Current Status Network VOIP Advanced Dial peer Config Manage Update System Manage e Web Update Web Update e FTP TFTP Update e Auto Provisioning Select file z or st i The device will reboot when update finish E Click the browse button find out the config file saved before or provided by manufacturer download it to the phone directly
5. Set examining interval of the server default is 60 seconds User Agent Set the user agent if have the default is VolP Phone 1 0 Signal Key Signal encryption Key v Media Key voice stream encryption Key v Local Port Local SIP signal port default as 5060 v Hotline Number Set hot line number of each line v MWI Number set SIP2 voicemail number Enable Conference Num conference ID Auto Detect Server Enable Disable keeps NAT of SIP alive If some server refuse to register with too short interval time and has no packets sending to device in private network to keep NAT alive user could set this function ON It need set the keep alive interval time less than the NAT server s Enable Keep Authentication Enable Disable Keep Authentication System will take the last authentication field which is passed the authentication by server to the request packet It will decrease the server s repeat authorization work if it is enable Enable Via rport Enable Disable system to support RFC3581 Via rport is special way to realize SIP NAT Enable PRACK Enable or disable SIP PRACK function suggest use the default ATCOM TECHNOLOGY CO LIMITED AT 620 User Manual ATC O wi config Long Contact Set more parameters in contact field Click To Talk Set click to Talk need practical software support Ban Anonymous Call Set to ban Anonymous Call Dial without Register Set call out by proxy without registration Enable Strict Pr
6. Specify Ring Volume grade G729 Payload Length Set G729 Payload Length Signal Standard Select Signal Standard G722 Timestamps 160 20ms or 320 20ms is available G723 Bit Rate 5 3kb s or 6 3kb s is available Default Ring Type Select signal standard VAD Select it or not to enable or disable VAD If enable VAD G729 Payload length could not be set over 20ms ATCOM TECHNOLOGY CO LIMITED AT 620 User Manual ATC O wi 4 11 VPN IP Phone gt A ATCOM Current Status Network VOIP Advanced Dial peer Config Manage Update System Manage e DHCP Server VPN Tunnel e NAT e Net Service e Firewall Qos e Digital Ma e STUN e Call Service e MMI Filter e Audio Settings e VPN UDP Tunnel Server Group ID IN d Server Area Code 2345 L2TP UDP Tunnel L2TP C Enable VPN APPLY this page is VPN setting page the IP phone support the VPN with UDP and L2TP protocol The parameters is as below VPN IP After VPN registered successfully VPN server will give an IP aggress to the terminal If there is a IP address shown on terminal except for 0 0 0 0 it means your VPN has registered UDP Tunnel VPN Server Addr register to the address of VPN server VPN Server Port Register to the port of VPN server Server Group ID The group ID of UDP VPN Server Area Code They are code of VPN server L2TP VPN Server Addr Register to the address of VPN server VPN User Name L2TP VPN username VPN P
7. VLAN tag 5 If disable the VLAN regardless to set the voice and data VLAN differentiated or not all packets will not take the VLAN tag if enable the DiffServ all packets will only take the DiffServ value 6 One must to notice enable the VLAN ID check enable that is default if enable gt Must to notice VLAN ID check Enable feature is default enable if enable it The phone will match the VLAN ID strictly When others VLAN ID mismatch with IP Phone the packets will discard Contrarily the phone will accept the packets with the distinct VLAN ID gt You must set the IP with static mode when you set VLAN otherwise can t obtain the IP in the VLAN and also cannot dial with point to point 4 6 Digital Map IP Phone Fa H ATCOM Current Status Network VOIP Advanced Dial peer Config Manage Update System Manage e DHCP Server Digital Map e NAT e Net Service e Firewall e Qos e Digital Ma e STUN e Call Service e MMI Filter e Audio Settings e VPN Digital Rule table A BDXXX IXXKXXKX 94 1 D KEN 6611 T4 Loo TERA Digit map is a set of rules to determine when the user has finished dialing AT620 support below digital map v End With Use as the end of dialing v Fixed Length The call will be sent out automatically when the length of the number you dial reaches the fixed one For example if you set number of 11 here when you dial 11 digits the call will be sent
8. e Phone Book e Syslog Config e Time Set Set Menu Password fae Menupassmora ln e Language Set D Logout Set e Reboot Set Keyboard Lock Keyboard Lock password Eable Keyboard Lock Set Users can add new account or delete and change existing account Set Menu Password Set menu of keypad password default is 123 Set KeyboardLock The default passwordis 123 It will take effect when you enable the keyboard lock The default setting is unlock if you press any key at this status the system will remind you to input password ATCOM TECHNOLOGY CO LIMITED AT 620 User Manual ATC O wi Greeting Message User name Confirm Set Backlight Timeout Set backlight time out if IP Phone has not press any operation to active within the settings value the backlight will off Set Greeting Message set the Greeting message on the LCD default is blank User Name set new account name User Level set new account level root can read and change setting general can only read v Password config password for new account v Confirm double confirm password If you want to make change on existing account select the account an click Modify or Delete General account can only modify or delete general account Keyboard Password config password that you use keyboard to access the menu must be in number ATCOM TECHNOLOGY CO LIMITED AT 620 User Manua
9. interval of the server default is 60 seconds User Agent Set the user agent if have the default is VolP Phone 1 0 Signal Key Signal encryption Key v Media Key voice stream encryption Key v Local Port Local SIP signal port default as 5060 v Hotline Number Set hot line number of each line v MWI Number Set SIP1 voicemail Number v Enable Conference Num conference ID Auto Detect Server Enable Disable keeps NAT of SIP alive If some server refuse to register with too short interval time and has no packets sending to device in private network to keep NAT alive user could set this function ON It need set the keep alive interval time less than the NAT server s Enable Keep Authentication Enable Disable Keep Authentication System will take the last authentication field which is passed the authentication by server to the request packet It will decrease the server s repeat authorization work if it is enable ATCOM TECHNOLOGY CO LIMITED v LN NN A S si S SS NN S S VV VSN Ss i E NS S AT 620 User Manual ATC O Uu Enable Via rport Enable Disable system to support RFC3581 Via rport is special way to realize SIP NAT Enable PRACK Enable or disable SIP PRACK function suggest use the default config Long Contact Set more parameters in contact field Click to Talk Set click to Talk need practical software support Ban Anonymous Call Set to ban Anonymous Call Dial Without R
10. no suttix 3 3 is the predecessor Then A could make the redial function via dialing 3 number of B 4 is the predecessor Then A could make the redial function via dialing 4 number of B User could name any predecessor like 3 4 if it is compliant with present dial rule 13 vport Vport makes more flexible calling application Eg It could forward a call from Line 1 to one account of Line 2 after configuring forward type and number line via web interface The forward could make either from Line 1 to Line 2 or Line 2 to Line 1 But the end user may not aware the configuration being made therefore probably the end user should be advised that it may cost with the forward function The forwarding could be done via either Line Key to select the line or dialing IP after calling under server It could be implemented by the following 4 ways 4 Point to Point Call Forward Make the configuration like ip port in the column of Forward Number Then it could make SIP call point to point with this IP and port in system User could select forward type accordingly Point to Point Blind Transfer Transfer the call via dialing IP directly Call Forward Call Transfer Blind Transfer Attended Transfer in different Line ATCOM TECHNOLOGY CO LIMITED AT 620 User Manual ATC O wi Make the configuration like sip username n in the column of Forward Number Then system would select Line N and make call accordingly SIP Line eg 0 1
11. out immediately Timeout Specify the timeout of the last dial digit The call will be sent after timeout v Prefix User define digital map represents the range of digit can be a range such as 1 4 or use comma such as 1 3 5 or use a list such as 234 ATCOM TECHNOLOGY CO LIMITED AT 620 User Manual ATC O Uu v x represents any one digit between 0 9 Tn represents the last digit timeout n represents the time from 0 9 second it is necessary Tn must be the last two digit in the entry If Tn is not included in the entry we use TO as default it means system will sent the number immediately if the number matches the entry Example gt 1 8 xxx All number from 1000 to 89999 will be sent immediately gt 9XXXXXXX 8 digits numbers begin with 9 will be sent immediately gt 911 Number 911 will be sent will be immediately gt 88xT4 3 digits numbers begin with 88with be sent after four seconds gt 6611x T4 holds four seconds send out if the number begins 6611 and five digits v Attention The above configuration can exist at the same time For example you enable as the signal of sending the call while set fixed length of 11 Either you press before the number reach 11 or dial 11 digital can send out the call 4 7 Stun IP Phone ATCOM Current Status Network VOIP Advanced Dial peer Config Manage Update System Manage e DHCP Server Stun Configuation e NAT e Net Service e Fi
12. press Update to save You can also update downloaded update file logo picture ring mmiset file by web 7 2 FTP TFTP Update IP Phone Sg H ATCOM Current Status Network VOIP Advanced Dial peer Config Manage Update System Manage e Web Update FTP TFTP Update e FTP TFTP Update e Auto Provisionin Server FTP TFTP server address It can be the format of IP address such as 192 168 1 1 or domain such as ftp domain com Meanwhile it Support sub directory such as 192 168 1 1 ftp config or ftp domain com ftp config v Username FTP user name TFTP no need v Password FTP password TFTP no need v File name the firmware or configuration file name that IP phone will search for in the server if leave it as blank the IP phone with search the file with the name of its MAC such as 000102030405 ATCOM TECHNOLOGY CO LIMITED AT 620 User Manual ATC O Uu Notice Users can revise the exported config file by themselves and import the config file with only modules for example if there is the SIP setting page in the config file the IP phone will only change SIP setting after import this file and leave other setting as not changed v Type upgrading type gt Application update update firmware gt Config file export export the current configuration to a FTP TFTP server gt Config fie import import configuration file from a FTP TFTP server gt Protocol choose server type FTP or TFTP 7 3 Au
13. your username of the voice mail is letters which you cannot input with the ATA then you use the number to stand for your username Voice mail text if IAX support voice mail config the domain name of your mail box here Echo test number If the platform support echo test and the number is test form the config the test number to replace the text format The echo test is to test the error status of terminals and platform Echo test text echo test number in text format Refresh time IAX refresh time Enable Register enable or disable register Enable 6 729 Using G 729 speech coding mandatory consultations SS NA ATCOM TECHNOLOGY CO LIMITED AT 620 User Manual ATC O wi 4 Advance 4 1 DHCP Server IP Phone a ATCOM Current Status Network VOIP Advanced Dial peer Config Manage Update System Manage e DHCP Server DHCP Configuration e NAT DHCP Leased Table e Audio Settings e VPN 192 168 10 30 1440 255 255 255 0 192 168 10 1 192 168 10 1 DHCP Leased 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 Leased IP Address the IP address which is assigned Client Hardware Address the IP address assigned and the MAC opposite of IP DHCP Lease Table Setting Lease Table Name Lease table name Lease Time DHCP server lease time Start IP Start IP of lease table En
14. 0 010 eee 19 P KEIER 20 2s NOWO oee E 20 2 1 WI ee de AE E E E 20 2 2 LAN CON ae Daio Ueba Dis peso dE asd ache adia cao nasal 22 3 NOU GE 23 3 1 OO ee 23 3 2 DL E 26 3 3 FP lu Le e 29 4 te Tee E 30 4 1 TAP SON CO asec E R E E 30 4 2 Va 31 4 3 Nera a cease deine RE EO SNIS E E CRER 34 4 4 lege 35 4 5 NOS seas snes SN ED Re DR 36 4 6 DIC PAIS APRE E OR ED 37 4 7 EE 38 4 8 EC UR e E 39 4 9 MINI 2 oe asewroensoasauonseeactseenteaeesaeraeniensanaacteoss E E EE reias esta ds 40 410 PUCIO SCTUMGS eseridir 4 4 11 MPN E DN RR NR IS SINA EDER DR E 42 Sa BIERG AAN AONE A AA A E 43 6 conng wt Le LC 45 Ta Eia e ihe EE 46 7 1 Wop UDU E 46 7 2 PRE a Ree 46 7 3 PUTO PROVISIONING iszscteiecnndscnsenonsasscenecsrsnseesesaceiwsdentaaisnensadesacseoesseend 47 8 YS EM gt 6 anna a 48 8 1 SEENEN JEE 48 8 2 PEO UN BOOK RAP DRE ERR A EE E RD 50 8 3 Sy SOC M 50 8 4 E Ria Ss PRERESI RR NR 52 8 5 SCH Ee BEE 53 8 6 Binet Lee WE 53 8 7 LO E 53 8 8 EEN 54 ATCOM TECHNOLOGY CO LIMITED AT 620 User Manual ATC O wi Ist AT 620 s Network Features 1 The View 2 Interfaces gt Power Output Power 12VDC 500mA gt WAN RJ45 port gt LAN RJ45 port A Electricity characteristic gt Specialty of electric output 12V 500mA DC gt The network connects 2 RJ 45 connect a WAN a LAN gt Headset jack RJ9 jack 2 ATCOM TECHNOLOGY CO LIMITED AT 620 User Manual ATC O wi 4 Softwa
15. 2 Or 0 0 0 0 0 0 0 1 0 0 0 2 255 255 255 255 which is compliant with former configuration Call Forward Call Transfer Blind Transfer Attended Transfer between SIP Line and Point to Pint It is compliant for the Call Forward Call Transfer Blind Transfer Attended Transfer between SIP Line and Point to Pint 14 Click to dial When User A accesses web interface and calls User B via clicking one link which is direct to B IP Phone of User A would ring Then call B automatically once User A picking up handset 15 SMS function gt Create new SMS 1 press MORE soft button 4 2 press SMS soft button 2 3 press NEW soft button 1 4 Edit SMS context and you can switch the input method by press such as ABC capital letters abc English letters 123 number input 5 When the edit is done press Send soft button 2 and input the receiver s phone number A press Sear soft button 1 to find the contact person in phonebook B directly input receiver s phone number C Use P2P method input IP address press 2 times to input For example if you send the SMS to the phone with IP address of 192 168 1 88 you will press 192 168 1 88 After inputting receiver s address press Send soft button 2 to send out gt SMS Check new SMS When there is a new SMS LCD will show New Message S 6 Press More soft button 4 7 Press MS soft button 2 LCD will display Number New Number old 8 If there is a new SMS
16. AT 620 User Manual ATC O wi AT 620 User Manual ISSUE 1 2 2009 10 22 ATCOM TECHNOLOGY CO LIMITED AT 620 User Manual ATC O wi lst AT 620 s Network Features 4 ls TIME VI N eege 4 2s Kier Le EE 4 3 Electricity ene ege e 4 4 SOLUNANO ene ee ea ea E s 5 5s NOON ORE E E RSRS CR eee 5 6 Management and Maintenance cccecccccsscccessseessssecessscessssesssees 6 Ts POCO OI ena o E ree Tne See ONTENTS Oe Te epee aaa 6 8 SOMITE S e BEE 6 Ox Operating Requirement cece cccssccesssccssssecesssecesssecesssecesssceseseeees 7 10 ELE AS US acest ve anise OD RD ND O 7 11 WTS CMU Ce VOU eege 7 2nd Feet installation Immestruction 8 3rd Keypad of IP Phone oii cccccccsecesssecesssecesssessssseeseeens 10 Ath Basic functions and operations eee cece eceestseeeeeeeeens 11 1 Answer the EE 11 25 WE EE 12 aK SOCO QI PR nem eee ee E A E E E EE E E A 13 4 Multiple ine TEE Die EE 13 5 Nano UDNE ee ru seeni Ea E Tener wee nD EE er 13 6 GAIN WM ANS DDD ET ERRO O RR RD EE T E EEE 14 T CTO WEE 15 8 3 Way e ein lui CC EE 15 9 AN TE ele OEE NEA A RD A PRN RA 15 10 SCHU a 16 11 Ro ais BP meee nn INN NR E O A 16 12 PEGI AN UMS e PARAR RR NR GR OR ERR RR 17 13 NOT esses is add epee cae ud ed da ed E a A N 17 14 CHUCK TO ON er IRD PED DIR RR RR UR RR NR RR 18 15 VA UII e E 18 ATCOM TECHNOLOGY CO LIMITED 16 Preload a SVN ON ease ect etsrs ds E 19 Lis cneck ne Phone EE 19 5th VAY 0s 1 0
17. OLOGY CO LIMITED AT 620 User Manual ATC O Uu Shows extension number and status There are three colors for LED red yellow and orange gt If the line is registered the LED shows yellow gt If the line is enable registered but register to server failed the LED shows orange If the line has income calling the LED shows red and flicker If the line is on the calling the LED shows red If the line disable for register the LED is off when there is the incoming call LED blinks The frequency is 500ms off 500ms on When have voicemail LED shows red and flicker The frequency is 1000ms off 1000ms on Navigation Allows users to navigate left right up down on the button Standby up and down shows the network information right Shows the lines information left shows the call record Speakerphone Pick up and hung up on the speakerphone mode when pick Mute button Mute the handset headset or speakerphone by press the Mute button this prevents the person on the active call form hearing what you or someone else in the room is saying To cancel the Mute function press the Mute button again If Mute the voice the LED is light on this button headset the LED button will light button will light Ath Basic functions and operations 1 Answer the calls When there is an incoming call AT620 will remind user with ringing There are 5 ways to answer the call A Answer by handset Pick up the handset and talk with the calle
18. When C s phone ring B hangs up the call with A the Led on B s Phone shows ols hang up 6 Creceives starts the call with A Remarks SIP lines are not available for choosing when call transfer gt Attended Transfer User A B C assume B is AT 620 Ip phone 1 When A Calls B and B receives 2 B presses soft button Xfer when A is calling 3 B dials C s number 4 After dialing C B Presses soft button Bxfe then transfers the call to C 5 C receives the phone starts the call with A 6 B presses soft button XFER directly starts to talk with A Meanwhile The LCD on B s phone shows pls hang up Remarks To carry out this function IP Phone must work with Call waiting and call transfer function meanwhile Sip server must support RFC3515 gt Alert Transfer User A B C assume B is AT 620 Ip phone 1 When A Calls B with B receives 2 B presses soft button Xfer when A Is calling 3 B dials C s number 4 After dialing C B Presses soft button Bxfe then transfers to C 5 When C s Phone ring B presses soft button XFER directly starts to talk with A Meanwhile The LCD on B s phone shows pls hang up 6 C receives the phone starts to talk to A Remarks To carry out this function IP Phone must work with Call waiting and call transfer function meanwhile Sip server must support RFC3515 ATCOM TECHNOLOGY CO LIMITED AT 620 User Manual ATC O wi ds Call
19. and 2 old SMS LCD will display 1 New 2 Old 9 Press OK soft button2 to enter SMS list if it s unread there will be a ATCOM TECHNOLOGY CO LIMITED 5th AT 620 User Manual ATC O wi NEW before it or else it has been read 10 Press up and down key in navigation keyboard to select the message and press ok soft button2 to read it 11 If you want to delete the SMS just press del soft button 1 after you select it Caution In SMS list you can press quit soft button to go to the upper menu Dial means dial to call sender directly when you are reading his SMS Edia means call the sender after edit his number Edit means editing the SMS context 16 Preload Password There are 2 models to set the authority of web accessing and command line Guest model and Admin model User could view and configure all items in Admin model While user couldn t change the SIP 1 2 and IAX2 configuration as well as server address and port but only access and view the information User would enter different model after input different user name and password e Guest Model 4 User Name guest 4 Pass word guest e Admin Model 4 User Name admin 4 Pass word admin 4 Keypad password 123 17 Check the Phone s IP Press the up or down navigation button to check the phone s IP address Web settings Enter AT 620 IP addresses in the web browser to go to the log on page and key in the username and password to access AT 620 setting page
20. apping table ATCOM TECHNOLOGY CO LIMITED AT 620 User Manual ATC O M 192 168 20 23 S002 S001 Shows the NAT UDP mapping table NAT Table Option 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 Oh Contig DM Table 192 166 1725 192 168 10 3 ww p e o ES DMZ Table Shows the outside WAN port IP address and the inside LAN port IP address Outside IP Set the outside wan port IP address of DMZ Inside IP Set the inside LAN port IP address of DMZ Click the Add button to add new table click the Delete button to delete the selected mapping table 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 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 ATCOM TECHNOLOGY CO LIMITED AT 620 User Manual ATC O Uu 4 3 Net Service IP Phone ATCOM Current Stat
21. assword L2TP VPN password UDPTunnel use the UDP to visit VPN L2TP use the L2TP to visit VPN Enable VPN Enable the VPN server you must choose UDP or L2TP type in advance ATCOM TECHNOLOGY CO LIMITED AT 620 User Manual ATC O Uu 5 Dial Peer IP Phone ATCOM Current Status Network VOIP Advanced Dial peer Config Manage Update System Manage Dial Peer Dial Peer Table ke ka ker TO ro ek kan no ro fes ua kan e ask eer eum po This functionality offers you more flexible dial rule you can refer to the following content to know how to use this dial rule When you want to dial an IP address the entry of IP addresses is very cumbersome but by this functionality you can set number 179 to replace 192 168 1 179 here When you want to dial a long distance call to China you need dial an country code 86 before local phone number but you can also dial number O instead of 86 after we make a setting according to this dial rule For example you want to dial 8675583018619 but you need dial only 075583018619 to realize your long distance call after you make this setting AT620 provide flexible dial rule with different dial rule configure user can easily implement the following function Replace delete or add prefix of the dial number Make direct IP to IP call Place the call to different servers according the prefix You can click Add to add a new dial rule Below is the detail setting of the dial r
22. ation Set Destination address This is optional config item If you want to ATCOM TECHNOLOGY CO LIMITED AT 620 User Manual ATC O Uu set peer to peer call please input destination IP address or domain name If you want to use this dial rule in SIP2 line you need input 0 0 0 2 in it If not config default sipl as 0 0 0 0 Port Set the Signal port the default is 5060 for SIP Alias Set alias This is optional config item If you don t set Alias it will show no alias Note There are four types of aliases 1 add xxx it means that you need dial xxx in front of phone number which will reduce dialing number length 2 all xxx it means that xxx will replace some phone number 3 del It means that phone will delete the number with length appointed 4 Rep It means that phone will replace the number with length and number appointed You can refer to the following examples of different alias application to know more how to use different aliases and this dial rule Call Mode Select difference 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 6 Config Manage IP Phone ATCOM Current Status
23. d IP End IP of lease table Network device connecting to the AT620 LAN port can dynamic obtain the IP in the range between start IP and end IP Net mask Net mask of lease table Gateway Default gateway of lease table DNS default DNS server of lease table Press add to apply will added DHCP lease table Lease Table Name Select name of lease table click the Delete button will delete the selected lease table from DHCP lease table DNS Relay Select DNS Relay the default is enable Click the Apply button to become effective DHCP Lease Table Shows the DHCP Lease Table the unit of Lease time is Minute 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 modifies the DHCP lease table you need save the configuration and ATCOM TECHNOLOGY CO LIMITED AT 620 User Manual ATC O wi reboot 4 2 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 Transfer Legality IP address DMZ config In order to make some intranet equipments support better service for extranet and make internal network security more effectively these equipments open to extranet need be separated from the other equip
24. data packets will add data VLAN ID data untagged means after using VLAN only VoIP packets will add voice VLAN ID Other data packets will not use VLAN DiffServ Enable Select it or not to Enable or disable DiffServ DiffServ Value Set DiffServ value the common value is 0x00 Voice 802 1P Priority Specify 802 1P Priority of voice signal data package Data 802 1P Priority Set 802 1p of data VLAN Non voip data such as http telnet ping etc will use this value to set VLAN package Voice VLAN ID Set VLAN ID of voice signal data package Data VLAN ID Set 802 1q of data VLAN ID Non VolP data such as http telnet ping etc will use this value to set VLAN package NOTICE 1 Enable VLAN if set Voice and Data VLAN differentiated as Undifferentiated all packets will use the Voice VLAN ID as the tag 2 Enable VLAN if set Voice and Data VLAN differentiated as tag differentiated and disable the DiffServ then system will not distinguish the voice and data all packets will use the Voice VLAN ID as the tag 3 Enable VLAN if set Voice and Data VLAN differentiated as tag differentiated ATCOM TECHNOLOGY CO LIMITED AT 620 User Manual ATC O Uu and enable the DiffServ then system will distinguish the voice and data and add the VLAN ID each other 4 Enable VLAN if set Voice and Data VLAN differentiated as date untagged then the packet of the signal and voice will use the voice VLAN ID as the tag but the data packets will not take the
25. de ping trace route telnet VV VV WM VV VV V V ATCOM TECHNOLOGY CO LIMITED AT 620 User Manual ATC O wi 6 Management and Maintenance Support safe mode and firmware updating under safe mode Support different level user management Configuration via web keyboard and command Support multi language LCD support Latin language system web support all languages and easy dynamic switch between different languages gt Firmware and configuration updating via HTTP FTP and TFTP gt Support system log and calling record gt Firmware firmware and configuration auto provision VV WV V 7 Protocol IEEE 802 3 802 3 u 10 Base T 100Base TX PPPoE PPP over Ethernet SIP RFC3261 RFC 2543 TCP IP Transfer Control Protocol Internet Protocol RTP Real time Transport Protocol RTCP RTP Control Protocol VAD CNG Telnet remote host access protocol DNS Domain Name Server TFTP Trivial File Transfer Protocol HTTP Hypertext Transfer Protocol FTP File Transfer Protocol VV VV VV VV VV V WV 8 Compliant Standard gt CE EN55024 EN55022 gt FCC part15 gt Comply with ROHS tn EU gt Comply with ROHS in China 68 Explanation The letter e is the first letter of environment and electronic The rim is a round with two arrow stands for recycle The number 20 stands for the years of environment protection Please note the years of environment ATCOM TECHNOLOGY CO LIMITED AT 620 User Manual
26. e LN NN NAN ATCOM TECHNOLOGY CO LIMITED AT 620 User Manual ATC O wi 8 5 Call Log IP Phone A N ATCOM Current Status Network VOIP Advanced Dial peer Config Manage Update System Manage e Account Manage Call LOG e Phone Book e Syslog Config e Time Set Call information e Call Lo e Language Set APR 16 14 39 sip 983018049 1 Start Time Display starts time of the outgoing record Last Time Display conversation time of the outgoing record Called Number Display the account protocol line of the outgoing record Notice It will cover existing automatically if the call log table has the new record 8 6 Language Set IP Phone TT ATCOM Current Status Network VOIP Advanced Dial peer Config Manage Update System Manage Account Manage Language selection Phone Book Syslog Config Time Set Call Log LANGUAGE SELECTION Language Set Logout Reboot Language Set English w Language Set Set the language of phone English is default Because we use 14px font on LCD so the Chinese and Korean language are not supported but only can be supported on web The default language is English if you need other language support please feel free to contact our sales 8 7 Logout IP Phone gt e S ATCOM Current Status Network VOIP Advanced Dial peer Config Manage Update System Manage Account Manage Logout Service Phone Book Syslog Config Time Set Call Log Language Set System L
27. e prefix and B s phone No C needed to set the dial peer with prefix code as follow To refer 1 as the set prefix code C could get the call from A to B by dialing 1 B 1 prefix could be freely set as long as no confliction with other dialing rules 11 Join call A could join in the conference call by input a prefix plus a phone No which is already in the conference A requested to set the prefix code for dial peer as follow To refer 2 as the set prefix code A could join in the conference by dial 2 plus the call No which is already in the conference 2 prefix could be freely set as long as no confliction with other dialing rules ATCOM TECHNOLOGY CO LIMITED AT 620 User Manual ATC O wi 12 Redial Unredial In order to being efficiently to contact the busy line A could use Redial to call B the busy line with setting prefix When B is free A could get through the call as usual When B is busy A could hang the phone with checking B s situation with every 60S by the set of prefix IP Phone of User A would ring and prompt picking up handset if B is available It would call B automatically once A picking up handset The call would get through as soon as had set being picked up at B A could dial the predecessor which set already add number of B to cancel the call before the phone automatic redialing if A is not available suddenly or don t want to call B anymore Sib 0 0 0 0 s060 SIP rep unredial
28. egister Set call out by proxy without registration Enable Strict Proxy Support the special SIP server when phone receives the packets sent from server phone will use the source IP address not the address in via field Forward Type Select call forward mode the default is Off 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 Always Incoming calls will be forwarded to the appoint phone directly The phone will prompt the incoming while doing forward Forward Phone Number Appoint your forward phone number Server Type Select the special type of server which is encrypted or has some unique requirements or call flows DTMF Mode Select DTMF sending mode there are three modes DTMF RELAY DTMF RFC2833 DTMF SIP INFO Different VolP Service providers may provide different modes RFC Protocol Edition Select SIP protocol version to adapt for the SIP server which uses the same version as you select For example if the server is CISC05300 you need to change to RFC2543 else phone may not cancel call normally System uses RFC3261 as default Transport Protocol Set transport protocols TCP or UDP Subscribe Expire Time Overtime of resending subscribe packet Suggest using the default config Conference Number config certain Conference call number Signal Encode enable si
29. enable voice data encryption Enable Session Timer enable rfc4028 to refresh the SIP sessions Answer With Single Codec only answer the call with a certain Codec Auto TCP enable TCP transmission protocol when the length of message exceed 1300 byte Enable URI Convert convert into 23 when sending URI Enable Display name Quote Set to make quotation mark to display name as the phone sends out signal in order to be compatible with server Enable GRUU Set to support GRUU Enable Subscribe Enable Subscribe Overtime of resending subscribe packet Suggest using the default config LN NN NN ATCOM TECHNOLOGY CO LIMITED AT 620 User Manual ATC O wi 3 3 lax2 Config IP Phone ae ATCOM Current Status Network VOIP Advanced Dial peer Config Manage Update System Manage Unregistered g I amp X2 Server Port 456 Local Port 45 Voice Mail Number Echo Test Number 60 Seconds Above is the IAX server configuration page AX Server Addr Register address of public IAX server v AX Server Port Register port of publiclAX server default port is 4569 Account Name Username of your SIP account Always the same as the phone number Account Password Password of your AX account Local port Signal port of local default port is 4569 Phone Number Phone number of your IAX account v Voice mail number If the IAX support voice mail but
30. ge e DHCP Server Firewall Configuration e NAT e Net Service e Qos e Digital Map e STUN e Call Service e MMI Filter Firewall Input Rule Table 1 Deny Joe 19216812 2552552550 192168103 255 255 255 0 Firewall Output Rule Table H Deny icmp 1921681060 255 2552550 192168170 255 255 255 0 in access enable Select it to Enable in access rule out_access enable Select it to Enable out_ access rule Firewall Input Rule Table Firewall input rule as the picture config is deny 192 168 1 2 ping 192 168 10 2 but ping 192 168 10 0 24 beside 192 168 10 3 is ok Firewall Output Rule Table Firewall output rule as the picture config is the phone ping 192 168 1 70 was deny Input iCutput Deny Permit e Audio Settings e VPN Sro adr o pes pn gene DesMask S Add Input output Specify current adding rule by selecting input rule or output rule Deny Permit Specify current adding rule by selecting Deny rule or Permit rule Protocol Type Filter protocol type You can select TCP UDP ICMP or IP Port Range Set the filter Port range Src Addr Set source address It can be single IP address network address complete address 0 0 0 0 or network address similar to 0 Dest Addr Set the destination address It can be IP address network address complete address 0 0 0 0 or network address similar to Src Mask Set the source address mask For example 255 255 255 255 means just point to one hos
31. gnal encryption Rtp Encode enable voice data encryption Enable Session Timer enable rfc4028 to refresh the SIP sessions Answer With Single Codec only answer the call with a certain Codec Auto TCP enable TCP transmission protocol when the length of message exceed 1300 byte Enable URI Convert convert into 23 when sending URI Enable Display name Quote Set to make quotation mark to display name as the phone sends out signal in order to be compatible with server Enable ATCOM TECHNOLOGY CO LIMITED AT 620 User Manual ATC O Uu GRUU Set to support GRUU Enable Subscribe Enable Subscribe Overtime of resending subscribe packet Suggest using the default config 3 2 SIP 2 IP Phone ATCOM Current Status Network VOIP Advanced Dial peer Config Manage Update System Manage SIP2 Configuation Basic Setting Register status Registered Proxy Server Address roxy Server Port roxy Username Leg Enable Register 4 APP Register Status SIP server registration status if succeed display Registered or else display Unregistered Server Address SIP server address support both IP address and domain name Server Port SIP server port default is 5060 Account Name SIP account name Phone Number SIP account phone number if leave it as blank no registration information will be sent out v Display Name Sh
32. is changed after effecting the configuration change the webpage will lose response former address so you must get to the webpage with new address 3 If the LAN IP address is happened to be the same as WAN IP which is allocated from DHCP server The LAN IP address will be changed automatically by adding 1 at the last digital 2 2 LAN Config IP Phone ATCOM Current Status Network VOIP Advanced Dial peer Config Manage Update System Manage e WAN Config LAN Configuration e LAN Confi Fe APPLY Parameter v LAN IP config LAN static IP ATCOM TECHNOLOGY CO LIMITED AT 620 User Manual ATC O Uu v Net mask LAN net mask DHCP Service enable LAN DHCP Server need to reboot to make it available NAT Network Address Translation Bridge Mode Select Bridge Mode or not If you select Bridge Mode the phone will no longer set IP address for LAN physical port LAN and WAN will join in the same network Click Apply the phone will reboot 3 VoIP 3 1 SIP1 IP Phone ATCOM Current Status Network VOIP Advanced Dial peer Config Manage Update System Manage SIP 1 Configuation Basic Setting Register status Registered Proxy Server Address Server Address 92 168 1 230 Proxy Server Port Register Status SIP server registration status if succeed display Registered or else display Unregistered Server Address SIP server address support both IP address and do
33. l ATC O wi 8 2 Phone Book IP Phone a ATCOM Current Status Network VOIP Advanced Dial peer Config Manage Update System Manage Account Manage Phone Book Phone Book Syslog Config Time Set Call Log Phonebook Table Language Set ie Wee LI me O Type Logout Reboot Default w Modify Phone Book v Phonebook Table shows phonebook detailed information v Add Phone Book add a new record in phonebook v Name nick name of a number when the call of this number comes in the LCD will show the name v Number phone number v Ring Type ring tone If you want to make change on existing account select the account an click Modify or Delete General account can only modify or delete general account Notice Maximum records of phone book is 500pcs 8 3 Syslog Config IP Phone a ERR ATCOM Current Status Network VOIP Advanced Dial peer Config Manage Update System Manage Account Manage Syslog Config Phone Book Syslog Config Time Set Syslog Set Call Lo SC Ft fm Tom MGR Log Level v Language Set Logout Reboot IAX2 Log Level None v APPLY Syslog is a protocol which is used to record the log messages with client server mechanism ATCOM TECHNOLOGY CO LIMITED AT 620 User Manual ATC O Uu 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
34. main name Server Port SIP server port default is 5060 Account Name SIP account name Phone Number SIP account phone number if leave it as blank no registration information will be sent out v Display Name Show the display name that you want to display on the phone of callee Support number and letter input Proxy Server Address Normally the Proxy server is the same as SIP server If they are different then fill in the correct information that provided by ISP Proxy Server Port Set your SIP server port Proxy Username Input your SIP register account name Proxy Password Input your SIP register password Domain Realm config SIP local domain If the server does not have special requirements for the local domain of SIP terminal the local domain can be the Same as SIP server domain The user can also leave it as blank the system will ATCOM TECHNOLOGY CO LIMITED AT 620 User Manual ATC O Uu take SIP server domain as the domain realm Enable Register Enable or disable registration Advanced SIP setting Advanced SIP Setting EE rar Cd CEE ER RER EE pronerwmsnsecase OQ CEE E KEE EE EE SSES Register Expire Time register expire time default is 600 seconds AT 620 will auto configure this expire time to the server recommended setting if it is different from the SIP server EE ee DEER EE ES parr ER EES EE EE EE v Auto Detect Server Interval Set examining
35. ments 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 equipments 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 ATCOM TECHNOLOGY CO LIMITED AT 620 User Manual ATC O wi lt GMZ area Intranet area The setting page as below IP Phone PA H ATCOM Current Status Network VOIP Advanced Dial peer Config Manage Update System Manage e DHCP Server NAT Configuration e NAT e Net Service rs PPTP ALG e QoS APPLY e Digital Map e STUN NAT Table e Call Service Inside IP Inside TCP Port Outside TCP Port e MMI Filter e Audio Settings Inside IP Inside UDP Port Outside UDP Port e VPN HAT Table Option IPSec ALG It is an encryption technology Select it to enable IPSec ALG the default is enable FTP ALG FTP is a service of connection layer which can transform intranet IP into extranet IP when intranet IP is sending out packet Select it to enable FTP ALG the default is enabling PPTP ALG Select it enable PPTP ALG the default is enable MAT Table Inside IP Inside TCP Port Outside TCP Port Shows the NAT TCP m
36. mum supports one incoming call when it is called when the second line calling the LCD will show the incoming telephone number The User can press the corresponding line key indicated by LED flicker or press soft button ANS to receive the second line call when two calls coming together press soft button SWIT to Switch Notice The phone must work with Call Waiting function when work for this feature 5 Hang up the phone 1 Headset hang up When use handset mode calling put back the handset to hang up 2 Hands free hang up When use hands free calling press soft button speaker phone to hang up 3 Earphone Hang up When use Earphone calling Press the soft button headset to hang up 4 Hang up one line call When 2 lines call simultaneous press soft button SWIT to choose the line which you want to hang up then press soft button to end the call In the mean time it will automatic switch to another line and continue call Moreover user can redial up or accept the second call Notice Hang up with is invalidation when only one line call ATCOM TECHNOLOGY CO LIMITED AT 620 User Manual ATC O wi 6 Call Transfer gt Blind Transfer User A B C assume B is AT 620 IP phone 1 When A Calls B and B receives 2 B presses soft button Xfer when A is calling 3 B dials C s number 4 After dialing C B Presses soft button xfer then transfers the call to C 5
37. net mask v MAC Address MAC of IP phone Current Gateway the IP address of the router B If your ISP provide you with the fixed IP address please choose static and fill in the correct information of IP Address Net mask Gateway Primary DNS etc If you do not know it please refer to your ISP provider or network management stuff The reference picture is as below IP Phone ATCQM Current Status Network VOIP Advanced Dial peer Config Manage Update System Manage e WAN Config WAN Configuation e LAN Config Parameters Static IP Address fixed IP address Net mask LAN net mask Gateway Gateway IP address DNS Domain input DNS domain name if it s provided Primary DNS Primary DNS address Alter DNS Alternative DNS address SN SS NS C when you use PPPoE to get IP address please select PPPoE and input ADSL account information as below picture ATCOM TECHNOLOGY CO LIMITED AT 620 User Manual ATC O wi IP Phone ATCOM Current Status Network VOIP Advanced Dial peer Config Manage Update System Manage e WAN Config WAN Configuation e LAN Config wan statas 00 0e 22 55 11 68 Parameters PPPoE Server sever name if the ITSP have no special requirements keep the ANY as default Username ADSL account user name Password ADSL account password Attention 1 After configuration setting please click Apply to effect the change 2 If the IP address
38. ny digital call with a certain head number For example 6 means any incoming number with the 6 as the first number will be refused gt 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 the settings as below 7049 means any incoming number is forbidden except 7049 Note End with DOT when set up the white list Limit List 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 4 9 MMI Filter IP Phone ATCOM Current Status Network VOIP Advanced Dial peer Config Manage Update System Manage e DHCP Server MMI Filter e NAT e Net Service e Firewall e Qos e Digital Map e STUN e Call Service e MMI Filter e Audio Settings e VPN 192 168 30 40 Sane Enae Ir E User could make some device own IP which is pre specified access to the MMI of
39. of wall mounting after fixing the two feet below Attention Please rotate the hook to the position as in picture with a coin or other tools D Disassemble the feet way Press the plate with word PUSH and pull the feet with the direction of arrow When the plate is pull out of the slot there will be a sound of pa you can take off the feet ATCOM TECHNOLOGY CO LIMITED AT 620 User Manual ATC O wi 3rd Keypad of IP Phone LCD Status SEP 14 11 15 Lines Soft buttons features Soft buttons Menu Navigation button Zes 8 Dial pad ooo Volume adjustment e buttons Voicemail Headset Mute Speaker phone button button button Describe of the buttons and Screen Soft button Shows available choices based on current phone function Status Shows the phone status if the phone is standby the LED is with light If there is income calling the LED will flicker If the phone is starting the LED is flicker if the phone is standby the LED is off If there is income calling the LED will flicker The frequency is 500ms off 500ms on When have voicemail LED shows red and flicker and the frequency is 1000ms off 1000ms on If the phone not obtain the IP address the LED is ON LCD Screen Display screen for the phone It shows the date time phone number incoming caller s ID if available line call status extension numbers and the soft button features ATCOM TECHN
40. ogout Logout 2gout Press the Logout button to Logout Phone Reboot Log out the configuration mode If you want to re configuration the phone need to input the user and password to login again ATCOM TECHNOLOGY CO LIMITED AT 620 User Manual ATC O Uu 8 8 Reboot IP Phone gt LAITT ATCOM Current Status Network VOIP Advanced Dial peer Config Manage Update System Manage Account Manage Reboot Phone Book Syslog Config Time Set Reboot Phone Call Lo Er Language Set Logout Reboot Reboot IP phone some setting needs to reboot to make it works Please always save config before reboot otherwise the setting will return to previous setting ATCOM TECHNOLOGY CO LIMITED
41. ow the display name that you want to display on the phone of callee Support number and letter input Proxy Server Address Normally the Proxy server is the same as SIP server If they are different then fill in the correct information that provided by ISP Proxy Server Port Set your SIP server port Proxy Username Input your SIP register account name Proxy Password Input your SIP register password Domain Realm config SIP local domain If the server does not have special requirements for the local domain of SIP terminal the local domain can be the Same as SIP server domain The user can also leave it as blank the system will take SIP server domain as the domain realm Enable Register Enable or disable registration Advanced SIP setting ATCOM TECHNOLOGY CO LIMITED AT 620 User Manual ATC O Wi Advanced SIP Setting Media Key O RFC Protocol Edition RFC3261 Iw Local Port 5060 E Transport Protocol Hotline Number subscribe Expire Time seconds hi Number Conference Number Loo Enable Keep Authentication Signal Encode O Auto Detect Server Rtg Encode Oo Enable Session Timer OO Anger With Single Coder Mo Enable Displayname Quote Enable Subscribe Register Expire Time register expire time default is 600 seconds AT 620 will auto configure this expire time to the server recommended setting if it is different from the SIP server Auto Detect Server Interval
42. oxy 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 Forward Type Select call forward mode the default is off gt Off Close down calling forward gt Busy Ifthe phone is busy incoming calls will be forwarded to the appointed phone gt No answer If there is no answer incoming calls will be forwarded to the appointed phone gt Always Incoming calls will be forwarded to the appoint phone directly The phone will prompt the incoming while doing forward Forward Phone Number Appoint your forward phone number Server Type Select the special type of server which is encrypted or has some unique requirements or call flows DTMF Mode Select DTMF sending mode there are three modes gt DTMF RELAY gt DTMF RFC2833 gt DTMF SIP INFO Different VolP Service providers may provide different modes RFC Protocol Edition Select SIP protocol version to adapt for the SIP server which uses the same version as you select For example if the server is CISCO5300 you need to change to RFC2543 else phone may not cancel call normally System uses RFC3261 as default Transport Protocol Set transport protocols TCP or UDP Subscribe Expire Time Overtime of resending subscribe packet Suggest to use the default config Conference Number config certain Conference call number Signal Encode enable signal encryption Rtp Encode
43. r If you want to hang up just put back the handset ATCOM TECHNOLOGY CO LIMITED AT 620 User Manual ATC O Uu D Hand Tree mode Press the hand free button in the phone and talk with callers by built in Micro phone and Speaker If you want to hang up please press the hand free button again C Answer by earphone Keep your earphone connected with the RJ 9 earphone jack when there Is an incoming call press the earphone button on the IP phone and talk with the caller If you want to hang up please press the earphone button again D Handset to hand free When you are phoning with the handset and want to phone with hand free mode please press the hand free button and put down the handset E Hand free mode to handset If you are phoning under hand free mode and want to change to speaker phone juts pick up the handset without press any buttons 2 Make Call A Use the handset Pickup the handset the LCD will show the current lines user could switch between linel and line2 by pressing the line button beside the LCD User can input the number with the keyboard and press to send the number When you hear the tones of du du with dialed number showed on the LCD the called s phone is ringing If the called answer the call the phone call is established and the LCD will show the calling time and the called s number B Answer the phone under band Tree mode Press the Speaker Phone button the LCD will show the current line
44. re gt Sip 2 0 RFC3261 gt Two lines SIP support AX2 gt STUN gt Jitter Buffer 200ms VAD CNG gt G 711A u 6722 6 723 6 729 Codec gt G 168 compliant 96ms echo cancellation gt Support SIP domain SIP authentication none basic MD5 gt Support inbound audio RFC2833 and SIP info DTMF transmission way gt SIP Call Forward Call transfer Call hold Call waiting 3 way talking Pickup Join call Redial Unredial Call Park Vport Click to dial gt Dial without register gt Support Hotline DND Do Not Disturb Blacklists Call Limitation DND Incoming list gt Dial peer calling rule IP to IP call SIP server conference Phone book with 500 records 100 answered call missed call for each Support HTTP FTP TFTP updating the configuration and firmware Syslog Answering machine Support SNTP client Telnet WEB visit terminal Support different level user management Support multi language LCD support Latin language system web support all languages gt soft button soft button 4 gt Support SMS VV VV VV VV WV 5 Network WAN LAN Support bridge or route mode Support base of NAT and NAPT Support PPPoE ADSL cable modem use for internet connecting Support VLAN DATA VLAN and VOICE VLAN Support DMZ Support L2ZTP VPN OpenVPN optional WAN support Primary and Alter function WAN support DHCP Client LAN support DHCP Server Qos support Diffserv Support Network command tool inclu
45. rewall e QoS e Digital Ma e STUN e Call Service STUN Set STUN NAT Transverse TRUE STUN Server Addr STUN Server Port 3478 e MMI Filter e Audio Settings e VPN STUN NAT Transverse STUN NAT Transverse status true or false STUN Server Addr configure stun server address STUN Server Port configure stun server port default 3478 STUN Effect Time stun detect NAT type interval time If NAT found a link inactive for a certain time it will close the link so you need to send a packet within a interval tome to keep the link alive Local SIP Port config local SIP port default as 5060Use Stun enable disable SIP STUN Attention AN ATCOM TECHNOLOGY CO LIMITED AT 620 User Manual ATC O Uu SIP STUN is used for NAT transverse When you config STUN server s address and port default 3478 and enable it then you can use the normal SIP server to make the IP phone transverse NAT 4 8 Call Service IP Phone SE ATCOM Current Status Network VOIP Advanced Dial peer Config Manage Update System Manage e DHCP Server Call Service Setting NAT Net Service Firewall Qos Digital Map e Audio Settings ge ae permeia o Apply v Hotline configure hotline number AT 620 immediately dials this number after hook off if it is set and the user can not dial any other number No Answer Time no answer call forward time setting No Disturb DND do no
46. 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 can not work Level 1 alert Your system has deadly problem Level 2 critical Your system has serious problem Level 3 error The error will affect your system working Level 4 warning There are some potential dangers But your system can work 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 send to Syslog is info debug level only can be displayed on telnet The items describe Server IP Syslog server IP address Server Port Syslog server port MGR Log Level config MGR log level SIP Log Level config SIPlog level IAX2 Log Level config IAX2log level Enable Syslog Enable Disable Syslog e Ce Ass ATCOM TECHNOLOGY CO LIMITED AT 620 User Manual ATC O Uu 8 4 Time Set Time setting Manual Timeset Rm Server type the IP address of time server Timezone select correct time zone in list box Timeout longest response time for SNTP Daylight Timeset daylight setting through manual Manual Timeset Time setting through manual Enable Daylight Daylight saving tim
47. s user could switch between linel and line2 by pressing the line button beside the LCD User can input the number with the keyboard and press to send the number When caller hear the tones of du du with dialed number showed on the LCD the called s phone is ringing If the called answers the call the phone call is established and the LCD will show the calling time and the called s number C Used phone book a Pick up the handset D Press Menu button and use the up and down keys to enter phonebook c Press OK to show the total amount in telephone d Press OK to enter the phone list and use up and down keys to find the contact person e When you find the certain contact person press OK to show the details f Press Edit to edit the number or press Dial to call ATCOM TECHNOLOGY CO LIMITED AT 620 User Manual ATC O wi 3 Speed dial It s method for the phone in standby mode to dial number immediacy The method is as below A Dial up the number in standby mode B Push soft button dail key or hang up directly to send the dial number C Push soft button to save the number in telephone directory 4 Multiple line dial up AT620 IP phone supports 2 Sip lines That means user can register on 2 different sip accounts simultaneity in the same IP phone The User can choose linel or line2 to switch dial up System default Sipl when dial up IP Phone be called AT 620 maxi
48. se missed call history 4 Choose the missed call record press OK soft button to browse the specific information of the record 5 Press Edai soft button to revise the records and press soft button dial to call this number gt Incoming call 1 Press the menu button 2 Press the navigation button to choose call history and then press OK button ATCOM TECHNOLOGY CO LIMITED AT 620 User Manual ATC O wi 3 Press the navigation button to choose incoming call press soft button OK 4 Press the navigation button to browse the incoming call record If there is no record the LCD screen display List is Empty gt Out coming call Method 1 1 Press Menu 2 Press up or down navigation key and select call history and press soft button OK 3 Select Outgoing call through up or down key and press soft button OK 4 Press up or down navigation button and check the received calls LCD will show List is Empty if there is no received incoming call Method 2 1 Press soft button Clog under standby status entering outgoing call list 2 Press up or down navigation button to read the received calls LCD will show List is Empty if there is no received incoming call 10 Call pickup Call pickup is simulated from Pickup function processes from IPPBX When A call B with no reply after ring tones C could pick up the call from A for B by inputting th
49. t 255 255 255 0 means point to a network which network ID is C type Des Mask Set the destination address mask For example 255 255 255 255 ATCOM TECHNOLOGY CO LIMITED ET S AT 620 User Manual ATC O wi means just point to one host if set to 255 255 255 0 means point to a network which network ID is C type 4 5 Qos IP Phone ATCOM Current Status Network VOIP Advanced Dial peer Config Manage Update System Manage DHCP Server QoS Configuration NAT Net Service Firewall C VLAN Enable Qos VLAN ID Check Enable Voice Data VLAN differentiated Undifferentiated Digital Map C DiffServ Enable DiffServ Value y Voice VLAN ID 256 0 4095 Data VLAN ID E Voice 802 1P Priority o om Data 802 1P Priority o STUN Call Service e MMI Filter e Audio Settings e VPN ADDI y APPLY VLAN Enable Before select it to enable VLAN you need enable Bridge mode in LAN config VLAN ID Check Enable Enable VLAN ID check by selecting it After enable VLAN ID check if VLAN ID of a data package is not the same with the phone s or a data package do not have VLAN ID the data package will be discarded Voice Data VLAN differentiated After enable VLAN system will set packets with different type of VLAN ID Undifferentiated means after using VLAN both voip packets and other data packets will use the voice VLAN ID tag differentiated means after using VLAN Vol P signal and voice packets will add voice VLAN ID and other
50. t disturb when there is an incoming call the caller will get the message that this line is not available but you it has no affection when you make outgoing call Ban Outgoing Enable this to ban outgoing calls Enable Call Transfer Enable Call Transfer by selecting it Enable Call Waiting Enable Call Waiting by selecting it Enable Three Way Call 3 way conference call Accept Any Call If select it the phone will accept the call even if the called number is not belong to the phone Auto Answer If select it the phone will auto answer when there is an incoming call P2P IP Prefix 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 Voicemail Number Set the voicemail number for each line Black List Set Add Delete Black list incoming call in these phone numbers will be refused It support below rules gt You add a certain number in it when this number call you it will be ATCOM TECHNOLOGY CO LIMITED AT 620 User Manual ATC O Uu refused gt Use x to represent any number For example 4xx means any incoming call with 3 digital and the first digital is 4 will be refused gt DOT means matching any arbitrary number digit for example any number with prefix 6 will be forbidden to dialed out A
51. the phone to config and manage the phone Add or delete the IP address segments that access to the phone Set initial ATCOM TECHNOLOGY CO LIMITED AT 620 User Manual ATC O Tvl IP address in the Start IP column Set end IP address in the End IP column and click Add to add this IP segment You can also click Delete to delete the selected IP segment Notice Do not set your visiting IP outside the MMI filter range otherwise you cannot logon through the web 4 10 Audio Settings IP Phone gt rs H ATCOM Current Status Network VOIP Advanced Dial peer Config Manage Update System Manage e DHCP Server Audio Settings e NAT e Net Service e Firewall e Qos e Digital Ma e STUN e Call Service e MMI Filter e Audio Settings DSP Configuration i g7 29 Fifth Codec Input volume 3 e VPN G729 Payload Length 20ms ze G722 Timestamps Default Ring Type Ype First Codec The fist preferential DSP codec G 711A u 6722 6 723 G 729 Second Codec The second preferential DSP codec G 711A u 6722 6 728 G 729 Third Codec The third preferential DSP codec G 711A u 6722 6 723 G 729 Forth Codec The Forth preferential DSP codec G 711A u 6722 6 723 6 729 Fifth Codec The fifth preferential DSP codec G 711A u 6722 6 723 G 729 Input Volume Specify Input MIC Volume grade Output Volume Specify Output receiver Volume grade Hands free Volume Specify Hands free Volume grade Ring Volume
52. to Provisioning IP Phone ATCOM Current Status Network VOIP Advanced Dial peer Config Manage Update System Manage e Web Update Auto Provisioning e FTP TFTP Update e Auto Provisioning Auto Update Setting Server Address Config Encrypt Key Protocol Type FTP 4 ser Disable hes Current Version the system will display the current version number need to modify the version id need to more than this number on the config file before auto provision update Server Address FTP TFTP server address v Username FTP server user name v Password FTP server password v Config File Name The name of configuration file Normally users leave it as blank the IP phone search for the file with the name same as its MAC in the server Config Encrypt Key The encrypt key of confirmation file v Protocol Type The protocol type that used for upgrading FTP TFTP and Http v Update Interval Time The interval time that the terminals search for new configuration file counted in hour v Update Mode auto provision mode A Disable not auto update B Update after reboot auto update after reboot C Update at time interval auto update after a certain time ATCOM TECHNOLOGY CO LIMITED AT 620 User Manual ATC O wi 8 System Manage 8 1 Account Manage IP Phone gt a ATCOM Current Status Network VOIP Advanced Dial peer Config Manage Update System Manage e Account Manage Account Manage
53. ule Phone Number The Number suit for this dial rule can be set as full match or prefix match Full match means that if the number user dialed is completely the Same as this number the call will use this dial rule Prefix match means that if prefix of the number that the user dials is the same as the prefix the call will use this dial rule to distinguish from the full match case you need to add T after the prefix number in the phone number setting Call Mode support SIP Destination optional call destination can be IP or domain Default is 0 0 0 0 in this case the call will be routed to the Public SIP server If you set the destination to 255 255 255 255 then the call will be routed to the private SIP server Also you can key other address here to make direct IP calls Port optional Configure the port of the destination default is 5060 in SIP Alias optional Set up the Alias We support four Alias as below Alias need to ATCOM TECHNOLOGY CO LIMITED emm mmm AT 620 User Manual ATC O Uu co work with the Del Length gt add xxx add prefix to the phone number can set to reduce the dial length gt all xxx replace the phone number with the xxx can use as speed dial function gt Del delete the first N numbers N is set in the Del Length gt rep xxx replace the first N numbers N is set in the Del Length For Example Use wants to place a call 8610 62281493 then you can set the phone number in the dial r
54. ule as O10T and set the Alias as rep 8610 and set the Del Length to 3 Then all calls begin with 010 will be changed to 8610 xxxxxxxx Suffix optional Configure suffix show no suffix if not set Instance description as picture 179 rule when you dial 179 the call with send to 192 168 1 179 suit for LAN application without set up a Sip server 3T rule If the call starts with 3 the first 3 will be deleted and the rest number with be sent to public SIP2 server 2T rule if the call starts with 2 the first 2 will be deleted and the rest number with be sent to IAX2 Server 123 rule Dial 123 and will send 8675583018049 to your server Used as speed dial function OT rule If the calls are begin with O the first O will be replacing by 86 Mean that if you dial 075583018049 and AT620 will send 8675583018049 to your server Add Dial Peer Phone Humber Destination optional Portcoptional Dial Peer Option ERR RR e o Phone number There are two types of matching conditions one is full matching the other is prefix matching In the full matching you need input your desired phone number in this blank and then you need dial the phone number to realize calling to what the phone number is mapped In the prefix matching you need input your desired prefix number and T then dial the prefix and a phone number to realize calling to what your prefix number is mapped The prefix number supports at most 30 digits Destin
55. us Network VOIP Advanced Dial peer Config Manage Update System Manage e DHCP Server Net Service e NAT e Net Service e Firewall QoS e Digital Ma e STUN e Call Service 40000 RTP Port Quantity 200 APPLY H modify HTTP or Telnet port you d better set it more than 1024 then restart e MMI Filter e Audio Settings e VPN HTTP Port set web browser port the default is 80 port if you want to enhance system safety you d better change it into non 80 standard port Example The IP address is 192 168 10 88 and the port value is 6090 the accessing address is http 192 168 10 88 6090 Telnet Port Set Telnet Port the default is 23 You can change the value into others Example The IP address is 192 168 1 88 the telnet port value is 6023 the accessing address is telnet 192 168 1 88 6023 RTP Initial Port Set the RTP Initial Port It is dynamic allocation RTP Port Quantity Set the maximum quantity of RTP Port the default is 200 Notice 1 You need save the configuration and reboot the phone after set this page 2 If you modify the port of Telnet and HTTP you would better set the value more than 1024 because the port value less than 1024 is system port reserved 3 if you set O for the HTTP port it will disable HTTP service ATCOM TECHNOLOGY CO LIMITED AT 620 User Manual ATC O wi 4 4 Firewall IP Phone ATCOM Current Status Network VOIP Advanced Dial peer Config Manage Update System Mana
Download Pdf Manuals
Related Search