AT-610P User Manual
Contents
1. 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 v 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 v Advanced SIP setting Advanced SIP Setting TTL Register Expire Time voip Phone 1 0 common w EES V Auto Detect Server emo 7 pese Dm mm 0 pesso D memo O per q ema Dm qu perene 0 femea ce E CE EENS Register Expire Time register expire time default is 600 seconds AT 610P will auto configure this expire time to the server recommended setting if it is different from the SIP server Auto Detect Server Interval 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 Sign2. the IP phone will only change SIP setting after import this file and leave other setting as not changed 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 ATCOM TECHNOLOGY CO LIMITED S AT 610P User Manual ATC O Tvl 7 3 Auto Provisioning IP Phone a ATCOM Current Status Network VOIP Advanced Dial peer Config Manage Update System Manage e Web Update Auto Provisioning e FTP TFTP Update s Auto Provisioning EE ET APPLY 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 Username FTP server user name Password FTP server password 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 af
3. Support Hotline DND Do Not Disturb Blacklists Call Limitation DND Incoming list gt VV VV V VV WV 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 5 Network VV VV VV WV WAN LAN Support bridge or route mode Support base of NAT and NAPT Support PPPoE ADSL cable modem use for internet connecting WAN support Primary and Alter function WAN support DHCP Client Qos support Diffserv Support Network command tool include ping trace route telnet 6 Management and Maintenance VV VV WM WV Support safe mode and firmware updating under safe mode Support different level user management Configuration via web keyboard and command Firmware and configuration updating via HTTP FTP and TFTP Support system log and calling record configuration auto provision ATCOM TECHNOLOGY CO LIMITED AT 610P User Manual ATC O vi 7 Protocol VV VV VV VV VV V WV 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 Tra
4. incoming call indicator A when the phone is in standby without incoming call the indicator is off B when the Ethernet connection is off the indicator is on C when there is the incoming call indicator blinks D when there is a new voicemail the indicator blinks Monday to Sunday indicator the correct day will be on while others will be off This is the multi function key p 1 ph O A In standby mode press Menu Enter to enter the Menu B In functional menu press Menu Enter to confirm selection Press info to check network status 16 7 press Pbook to enter the phone book and edit or delete 1 bech OO bech NO Press History button to check call history including incoming calls outgoing calls and missed calls Exit button A when you re dialing press exit to delete the dialed number and return to and you can dial a new number B in functional menu press EXIT to exit current menu and return to the upper ATCOM TECHNOLOGY CO LIMITED 4th AT 610P User Manual ATC O ivi conference key when you want to realize conference call press CONF key and key in the third party number Transfer key Press Hold to hold the call when you are talking 23 Press mute you can hear the voice from the other end but he can not hear you speaking 24 Voice volume key A press to turn up the voice volume RB press to turn up the voice volume Numeral leys Hand free button to ente
5. need practical software support Ban Anonymous Call Set to ban Anonymous Call Dial Without Register 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 ATCOM TECHNOLOGY CO LIMITED AT 610P User Manual ATC O vi which uses the same version as you select For example if the server is CISC0O5300 you need to change to RFC2543 else phone may not cancel call normally System uses RFC3261 as default v Transport Protocol Set transport protocols TCP or UDP Subscrib
6. E beer O Advanced Set ATCOM TECHNOLOGY CO LIMITED v v AT 610P User Manual ATC O Tvl 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 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 take SIP server domain as the domain realm v Enable Register Enable or disable registration Advanced SIP setting Advanced Sel Sie ped m CEE O Poet ao E ro A D ma gt E a E g A E E L CEE E EE CEC ER Register Expire Time register expire time default is 60
7. Network 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 AR ATCOM Current Status Network VOIP Advanced Dial peer Config Manage Update System Manage WAN Configuation mansa O O 00 03 04 05 09 28 WAN Setting sac ez rO APPLY Parameters v Active IP IP phone s address Current Net mask network net mask v MAC Address MAC of IP phone ATCOM TECHNOLOGY CO LIMITED AT 610P User Manual ATC O ivi 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 ATCOM Current Status Network VOIP Advanced Dial peer Config Manage Update System Manage WAN Configuation FER o OO MAC Address 00 03 04 05 09 28 been besen Parameters v 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 a
8. the outgoing record Notice It will cover existing automatically if the call log table has the new record 8 6 Logout IP Phone ATCOM Current Status Network VOIP Advanced Dial peer Config Manage Update System Manage Logout Service e Account Manage es Phone Book e Syslog Config e Time Set E Call Log System Logout e bo e Press the Logout button to Logout Phone Log out the configuration mode If you want to re configuration the phone need to input the user and password to login again 8 7 Reboot E IP Phone ATCOM Current Status Network VOIP Advanced Dial peer Config Manage Update System Manage Reboot e Account Manage e Phone Book Syslog Confi Reboot Phone gt ysiog Contig E EE ah Press the Reboot button to reboot Phone e Call Log e 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
9. 0 seconds AT 610P will auto configure this expire time to the server recommended setting if it is different from the SIP server ATCOM TECHNOLOGY CO LIMITED v MSN SS SS VS SS Ss S SS RK LN NN S S VV WV SN AT 610P User Manual ATC O vi Auto Detect Server Interval 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 Media Key voice stream encryption Key Local Port Local SIP signal port default as 5060 Hotline Number Set hot line number of each line 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 config Long Contact Set more parameters in contact field Click to Talk Set click to Talk
10. 23 4 2 GCN e E 26 4 3 Rn 27 4 4 A SNCS E 28 4 5 NEE EE 29 ATCOM TECHNOLOGY CO LIMITED 5 BIERG 30 6 6 a i te LC 33 Ss Bjo o LEE 33 7 1 W PUDO O E 33 L PER SE Ee el 34 73 PUTO PROVISI OMNIA G sicoecsccavcnnnsasewsceasstesoansencesieossaciassessaiiewenssisainetsanseaers 35 8 SNE Le TEE 36 8 1 EE lte 36 8 2 PHONG BOOK E 37 8 3 DY SIO e LEE 37 8 4 DR pa Le EE 39 8 5 SCHU EO ME 40 8 6 Ree soca assa Saad shan EE EG ES CEIA E ANAE 40 8 7 Ee 40 ATCOM TECHNOLOGY CO LIMITED AT 610P User Manual ATC O ivi Ist AT 610P 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 Specialty of electric output 12V 500mA DC The network connects 2 RJ45 connect a WAN a LAN Headset jack RJ9 jack 2 Support PoE VN NV V 4 Software gt Sip 2 0 RFC3261 gt Two lines SIP support IAX2 ATCOM TECHNOLOGY CO LIMITED AT 610P User Manual ATC O ivi gt VV VV NON STUN Jitter Buffer 200ms VAD CNG G 711A u 6722 6 723 G 729 Codec G 168 compliant 96ms echo cancellation Support SIP domain SIP authentication none basic MD5 Support inbound audio RFC2833 and SIP info DTMF transmission way SIP Call Forward Call transfer Call hold Call waiting 3 way talking Pickup Join call Redial Unredial Call Park Vport Click to dial gt gt Dial without register
11. AT 610P User Manual ATC O ivi AT 610P User Manual ISSUE 1 0 2009 9 11 ATCOM TECHNOLOGY CO LIMITED AT 610P User Manual ATC O ivi lst AT 610P s Network Features 4 ls DR 4 2 year e 4 3 Segelen leg E Le 4 Ge EE EE 4 Ss IS lt a rea AA AAE RR O RR AEA ETRE 5 6 Management and Maintenance 0 cece cece cccsssccessecssssecesssecesssseeseees 5 ER Steg ele EE 6 8 Compliant Standard eese Sad Eras Sia EEE ars Cais ia Estas ab atas d s 6 9 Operating Requirement EE 6 10 Sie dite fl NS sos aigasmondosaaiaddspndissdsdsadaidc medindo polca dicas sdasa cd aids nais do atadas ieeetoacaepectetisoecs 6 II He ein EE 7 2nd Feet installation Immestruction 7 3rd Keypad of IP Phone oie cceccccceseccsssecssssecsssseesssseseeeens 10 Ath Basic functions and operations eee 12 1 AN WE ENC EE 12 2s WH e E 13 3 ake 6 Ojo dida eine EE 13 4 SCHU ee E SAR RED O RR O 13 Ss S WaV CONIErence Call EE 14 6 CHON EE 14 S CoN Da TOV e E E E EE E EAT I 14 8 SC UN POG IO EE 15 9 POR UM EE 15 10 Kee I Bt e e EE 15 11 Eege 16 12 S el eh e o f EE 17 13 Pron ad P WO cenen E nn asda 17 14 check the PRONES oh agate meen en E EEEN EEA 17 5th WOD selo 6 oora E E EO 17 ls TIE PNG SCOLO osisticiasemisidenuisia fedesisddo o iiinfocadsnidegadido er oi nia oon iieri asiarren 18 2s IW OU GENRE RPPN nace stance OR ND RD RR 18 3 VOIP areca cas REPENTE SEE RAS OR RE 20 4 PAO ne NPR A RM O RR O 23 4 1 CH GE
12. AT 610P User Manual ATC O vi 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 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 5 Dial Peer IP Phone ATCOM Current Status Network VOIP Advanced Dial peer Config Manage Update System Manage Dial Peer Dial Peer Table 1921681179 i eg ES SE E fem Sharks fem fo 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 distan
13. GY CO LIMITED AT 610P User Manual ATC O ivi 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 610P User Manual ATC O vi C Repeat Aand B Itis the picture 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 610P User Manual ATC O Uu 3rd Keypad of IP Phone 12345 6 7 89101112 13 T eee amp d D i ES O A G A HUOT 24 25 26 29 28 27 Sketch Map of Phone Description 1 Network status indicator A when the network is OK it s on B when the network can not connect it blinks Network mode indicator A Number 3 stands for DHCP when the IP phone get IP address in DHCP Mode the indicator is on or else it blinks B Number 5 stands for Static when the IP phone get IP add
14. 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 ATCOM TECHNOLOGY CO LIMITED AT 610P User Manual ATC O vi 6 Config Manage IP Phone ATCOM Current Status Network VOIP Advanced Dial peer Config Manage Update System Manage Config Manage Save Configuration Press the Save button to save the configuration files Save Backup Config Save all Network and YolP 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 7 Update 7 1 Web Update IP Phone ATCOM Curr
15. al Key Signal encryption Key Auto Detect Server Interval To Oy CC Conterence Number Enable Keep Authentication Ee Lesen conterere uneer Et Koen Atherton puto ett Server UI ATCOM TECHNOLOGY CO LIMITED AT 610P User Manual ATC O ivi v Media Key voice stream encryption Key v Local Port Local SIP signal port default as 5060 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 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 pr
16. ble 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 refused gt Use x to represent any number For example 4xx means any incoming ATCOM TECHNOLOGY CO LIMITED AT 610P User Manual ATC O ivi 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 Any 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 wh
17. ce call after you make this setting AT610P 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 rule ATCOM TECHNOLOGY CO LIMITED AT 610P User Manual ATC O vi 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 co work with the Del Length gt add xxx add
18. ddress Alter DNS Alternative DNS address 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 610P User Manual ATC O vi IP Phone Ge ATCOM Current Status Network VOIP Advanced Dial peer Config Manage Update System Manage WAN Configuation 192168 1 48 255 255 255 0 192 168 1 254 00 03 04 05 09 28 20090625 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 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 1f 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 3 VoIP IP Phone Pa D ATCOM Current Status Network VOIP Advanced Dial peer Config Manage Update System Manage Public SIP Configuation asic Setting Register status Registered Proxy Server Address Lo Jo o Server Address 1 92 168 1 230 Proxy Server Port LJJINHD presa fumo bm 7 isplay Name CE
19. e Expire Time Overtime of resending subscribe packet Suggest using the default config v Conference Number config certain Conference call number Signal Encode enable signal encryption v Rtp Encode enable voice data encryption v Enable Session Timer enable rfc4028 to refresh the SIP sessions v Answer With Single Codec only answer the call with a certain Codec v Auto TCP enable TCP transmission protocol when the length of message exceed 1300 byte Enable URI Convert convert into 23 when sending URI v 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 4 Advance 4 1 SIP IP Phone gt a ATCOM iene A Sacer Public SIP Configuation Basic Setting P rver Port P Server Address roxy Server Port LY Phone Number 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 v Account Name SIP account name ATCOM TECHNOLOGY CO LIMITED AT 610P User Manual ATC O vi
20. eck 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 AX2 Log Level config AX2log level Enable Syslog Enable Disable Syslog KC le SS Ss ATCOM TECHNOLOGY CO LIMITED AT 610P User Manual ATC O ivi 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 time LN NN NAN ATCOM TECHNOLOGY CO LIMITED AT 610P User Manual ATC O ivi 8 5 Call Log IP Phone ATCOM Current Status Network VOIP Advanced Dial peer Config Manage Update System Manage Call LOG e Account Manage hd Phone Book Call information Syslog Gonti Start me LastTime Called Number e Time Set e Logout e Reboot 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
21. ent 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 I Click the browse button find out the config file saved before or provided by manufacturer download it to the phone directly press Update to save You can ATCOM TECHNOLOGY CO LIMITED AT 610P User Manual ATC O ivi also update downloaded update file logo picture ring mmiset file by web 7 2 FTP TFTP Update IP Phone 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 Provisioning ation update 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 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 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
22. epresents 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 610P User Manual ATC O ivi 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 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 3 Stun IP Phone Current Status Network VOIP Advanced Dial peer Config Manage Update System Manage Stun Configuation e Digital Map e STUN e Call Service e Audio Settings STUN NAT Transverse STUN Server Addr STUN Server Port 3478 STUN Effect Time 50 STUN NAT Transverse STUN NAT Transverse s
23. ference Number config certain Conference call number Signal Encode enable signal 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 GRUU Set to support GRUU Enable Subscribe Enable Subscribe Overtime of resending subscribe packet Suggest using the default config e Ss e SS NS SS 4 2 Digital Map IP Phone Current Status Network VOIP Advanced Dial peer Config Manage Update System Manage Digital Map e Digital Map e STUN e Call Service e Audio Settings Digit map is a set of rules to determine when the user has finished dialing AT610P 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 out immediately v Timeout Specify the timeout of the last dial digit The call will be sent after timeout v Prefix User define digital map r
24. 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 Destination Set Destination address This is optional config item If you want to set peer to peer call please input destination IP address or domain name If you want to use this dial rule 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
25. ite 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 5 Audio Settings IP Phone Current Status Network VOIP Advanced Dial peer Config Manage Update System Manage Audio Settings e Digital Map e STUN e Call Service e Audio Settings DSP Configuration First Codec g711Ulaw64k ia g 1iAlawb4k w Third Codec g729 g723 arsane E EE 160 20ms 6 3kb s First Codec The fist preferential DSP codec G 711A u 6722 6 723 6 729 Second Codec The second preferential DSP codec 6 711A u 6722 6 728 6 729 Third Codec The third preferential DSP codec G 711A u 6722 6 723 6 729 Forth Codec The Forth preferential DSP codec G 711A u 6722 6 723 6 729 ATCOM TECHNOLOGY CO LIMITED
26. ng up handset 13 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 Pass word guest e Admin Model 4 User Name admin 4 Pass word admin 14 Check the Phone s IP Press the up or down navigation button to check the phone s IP address Web settings Enter AT 610P IP addresses in the web browser to go to the log on page and key in the username and password to access AT 610P setting page Default username and password is Administrator Username admin password admin User Username guest Username guest ATCOM TECHNOLOGY CO LIMITED AT 610P User Manual ATC O vi 1 Current state IP Phone GE 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
27. nsfer Protocol HTTP Hypertext Transfer Protocol FTP File Transfer Protocol 8 Compliant Standard VN V WV CE EN55024 EN55022 FCC part15 Comply with ROHS in EU Comply with ROHS in China So 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 protection is not discarding year nor usage life H Operating Requirement Operation temperature O to 40 C 32 to 104 F Storage temperature 30 to 65 C 22 to 149 F Humidity 10 to 90 no dew Packing List AT 610P IP phone Power adaptor output 12v 500mA Manual CD ATCOM TECHNOLOGY CO LIMITED a AT 610P User Manual ATC O ivi 11 Installation Use Ethernet cable to connect AT 610P 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 610P the default username and password is admin admin for administrator and guest guest for guest Set up page for VolP user only bwitch AT 610 AT 610 AT 610 AT 610 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 ATCOM TECHNOLO
28. o LCD display RECEIVED 2 Press Menu Enter to display received call records 3 press Vol or Mol browse received call records 4 Choose the received call record press Redial button to call this number If there is no record the LCD screen display List is Empty gt Out coming call 1 Press History button and press Vol or Vol to LCD display DIALED 2 Press Menu Enter to display out coming call records 3 press Vol or Vol browse out coming call records ATCOM TECHNOLOGY CO LIMITED AT 610P User Manual ATC O ivi 4 Choose the received call record press Redial button to call this number If there is no record the LCD screen display List is Empty 8 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 the 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 H 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
29. oxy 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 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 ATCOM TECHNOLOGY CO LIMITED AT 610P User Manual ATC O vi the default config Con
30. 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 ATCOM Current Status Network VOIP Advanced Dial peer Config Manage Update System Manage Account Manage Syslog Config e Phone Book e Syslog Config e Time Set e Call Log e Logout e Reboot Syslog is a protocol which is used to record the log messages with client server mechanism Syslog server receives the messages from clients and classifies them based on priority and type Then these messages will be written into log by some rules ATCOM TECHNOLOGY CO LIMITED AT 610P User Manual ATC O ivi 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 ch
31. 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 rule as 010T 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 AX2 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 AT610P will send 8675583018049 to your server ATCOM TECHNOLOGY CO LIMITED AT 610P User Manual ATC O ivi Add Dial Peer Phone Humber Destination optional Portcoptional Dial Peer Option sl ec S Phone number There are two types of matching conditions one is
32. r 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 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 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 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 ATCOM TECHNOLOGY CO LIMITED 5th AT 610P User Manual ATC O ivi 12 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 picki
33. r hand free mode 27 Dial button when you finish dialing the telephone number press this button to Wl send out the phone number Del button is used to delete a single number or letter Headset button is used to enter headset answering mode Basic functions and operations 1 Answer the calls When there is an incoming call AT610P 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 caller If you want to hang up just put back the handset D Hand free 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 ATCOM TECHNOLOGY CO LIMITED AT 610P User Manual ATC O vi 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 can input the numbe
34. r 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 D Answer the phone under band Tree mode Press the Speaker Phone button the LCD will show the current lines 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 phone D Press Pbook button c Press Menu Enter to enter the phone list and use Vol or Vol keys to find the contact person d When you find the certain contact person press Menu Enter to show the details e Press Edit to edit the number or press Dial to call A 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 Call Transfer gt Unattended transfer If A is using AT510 talking with B B want to speak to C A just p
35. ress XFER and dial C s number gt Attended transfer ATCOM TECHNOLOGY CO LIMITED AT 610P User Manual ATC O vi Only SIP support attended Transfer If A is using AT510 talking with B B want to speak to C A just press Hold and dial C s number to ask whether he can answer the call from B C agree then press Hold to talk with B and press XFER to transfer the call 5 3 way conference call Enable Three Way Call If A is using AT 510 talking with B and B want to make conference call with A and B A just press Hold and dial C s number Then press CONF to initiate conference call 6 Call Hold User can hold the current call by pressing soft button Hold And by pressing soft button Hold again 7 Call History AT 610P 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 display the Missed call icon and the status LED is 2s on 500ms off 2 Press history button press Vol or Vol to LCD display Missed 3 Press Menu Enter to display the records of Missed call 4 press Vol or Vol browse missed call history 5 Choose the missed call record press Redial button to call this number gt Incoming call 1 Press History button and press Vol or Vol t
36. ress in Static mode the indicator is on or else it blinks ATCOM TECHNOLOGY CO LIMITED AT 610P User Manual ATC O vi C Number 7 stands for PPPoE when the IP phone get IP address in PPPoE mode the indicator is on or else it blinks 2 4 6 Answering mode indicator A Number 2 stands for answering call with handset RB Number 4 stands for answering call in hand free mode C Number 6 stands for answering call with headset Mute indicator when the phone is muted this indicator is on when the muted mode is canceled the indicator will be off 11 Protocol indicator A Number 9 stands for SIP gt When you enable register and successfully register to SIP server indicator is on gt when you enable register but do not successfully register to server indicator blinks gt when you disable register this indicator is off RB Number 11 stands for IAX protocol gt When you enable register and successfully register to JAX server indicator is on gt when you enable register but do not successfully register to server indicator blinks when you disable register this indicator is off p Missed call indicator when there is the missed call this indicator will be After you checked all the missed calls this indicator will turn off bech Voicemail indicator when there is new voice mail this indicator will be on after you checked all the voicemail this indicator will turn off p lt
37. tatus 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 SIP STUN is used for NAT transverse When you config STUN server s v v v v ATCOM TECHNOLOGY CO LIMITED AT 610P User Manual ATC O ivi address and port default 3478 and enable it then you can use the normal SIP server to make the IP phone transverse NAT 4 4 Call Service IP Phone Current Status Network VOIP Advanced Dial peer Config Manage Update System Manage Call Service Setting CE ER E femme 0 fem ja pena fa e Digital Map STUN dores peer retro e Call Service e Audio Settings v Hotline configure hotline number AT 610P 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 not 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 Ena
38. ter reboot auto update after reboot C Update at time interval auto update after a certain time SS NAN S ATCOM TECHNOLOGY CO LIMITED AT 610P User Manual ATC O ivi 8 System Manage 8 1 Account Manage IP Phone ba ATCOM Account Manage e Account Manage e Phone Book e Syslog Config e Time Set Set Keyboard Password e Logout e Reboot Users can add new account or delete and change existing account Keyboard Password config password that you use keyboard to access the menu must be in number v User Name set new account name v 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 ATCOM TECHNOLOGY CO LIMITED AT 610P User Manual ATC O vi 8 2 Phone Book IP Phone ATCOM Current Status Network VOIP Advanced Dial peer Config Manage Update System Manage Phone Book e Account Manage e Phone Book S slog Conti Phonebook Table e Time Set e Call Log index JL me Etter D e Logout e Reboot Add Phone Book kel Modify Phone Book vi sunny w v Phonebook Table shows phonebook detailed information v Add Phone Book add a new record in
39. 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 10 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 ATCOM TECHNOLOGY CO LIMITED AT 610P User Manual ATC O ivi anymore sq 0 0 0 0 SCH SIP rep unredial 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 11 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 eithe
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