Home

AIP-311 VOIP PHONE User manual

Contents

1. Authentication Shows 1f the phone has been registered the SIP server or not or so show Unapplied Set the server name Input your SIP server address Set your SIP server port Input your SIP register account name Input your SIP register password Input the phone number assigned by your VoIP service provider Phone will not register 1f there 1s no phone number configured Set the display name Set proxy server IP address Usually Register SIP Server configuration is the same as Proxy SIP Server But 1f your VoIP service provider give different configurations between Register SIP Server and Proxy SIP Server you need make different settings Set your Proxy SIP server port Input your Proxy SIP server account Input your Proxy SIP server password Set the sip domain if needed otherwise this VoIP phone will use the Register server address as sip domain automatically Usually it is same with registered server and proxy server IP address Start to register or not by selecting it or not Set expire time of SIP server register default is 60 seconds If the register time of the server requested is longer or shorter than the expire time set the phone will change automatically the time into the time recommended by the server and register again Set examining interval of the server default is 60 seconds Set the user agent if have the default is VoIP Phone 1 0 Set the key for signal encryption Set the key for RTP
2. G 729 G 726 Third Codec The third preferential DSP codec G 711A u G722 G723 G 729 G 726 AMR Forth Codec The forth preferential DSP codec G 711A u G722 G723 G 729 G 726 AMR Fifth Codec The fifth preferential DSP codec G 711A u G722 G723 G 729 G 726 AMR Sixth Codec The fifth preferential DSP codec G 711A u G722 G723 G 729 G 726 AMR Seventh Codec The seventh preferential DSP codec G 711A u G 722 G 723 G 729 G 726 AMR AMR Payload Type AMR Payload Type Handdown Time Default Ring Type Input Volume Output Volume Hands free Volume Specify the least reflection time of Handdown the default is 200ms Set up the ring by default Specify Input MIC Volume grade Specify Output receiver Volume grade 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 VAD Select it or not to enable or disable VAD If enable VAD G729 Payload length could not be set over 20ms DTMF Payload Type Set up DTMF payload type 4 3 4 2 Call Service In this web page you can configure Hotline Call Transfer Call Waiting 3 Ways Call Black List 38 white list Limit List and so on PHONE cano service SS A ANTES Call Service Setting Auto Handdown Time 3 seconds aL Server Black List A laa A
3. connection with SEM server Convert to 23 when send the URI Set call out by proxy without registration Set to ban Anonymous Call Support DNS looking up with _sip udp mode Select call forward mode the default is Off o Off Close down calling forward Oo Busy If the phone is busy incoming calls will be forwarded to the appointed phone O No answer If there is no answer incoming calls will be forwarded to the appointed phone O Always Incoming calls will be forwarded to the appoint phone directly The phone will Prompt the incoming while doing forward Appoint your forward phone number Select the special type of server which is encrypted or has some unique requirements or call flows Select DTMF sending mode there are three modes DTMF_RELAY Oo DTMF_RFC2833 Oo DTMF_SIP_INFO Different VoIP Service providers may provide different modes Select SIP protocol version to adapt for the SIP server which uses the same version as you select For example if the server is CISCO53300 you need to change to RFC2543 else phone may not cancel call normally System uses RFC3261 as default Set transport protocols TCP or UDP Set Anonymous call out safely Support RFC3323and RFC3323 Overtime of resending subscribe packet Suggest using the default config Set to use sever conference Input the number of the server s voice mail box Set click to Talk need practical software support Y Signal Encode RTP Enc
4. 1 Prepare the network s parameters first such as IP Address Net mask Default Gateway and DNS server IP address If you don t know this information please contact the service provider or technician of network 2 Press then press ES twice chooses Network Press OK or Soft2 Enter LCD screen will display WAN 3 Press OK or Soft2 Enter then choose Static 4 Press Soft1Edit and screen will show IP then press Soft1 Del to delete Input your IP address and press Soft2 Save to save what you input After Saved shown the screen will jump to show the Net mask information 5 Press Soft1 Del to delete Input your Net mask and press Soft2 Save After Saved shown the screen will jump to show the Gateway information 6 Press Soft1 Del to delete Input your gateway and press Soft2 Save After Saved shown the screen will jump to show the DNS information 7 Press Soft Del to delete Input your DNS server address and press Soft2 Save After Saved shown the screen will return to show IP information 8 Press Soft3 Quit once the screen shows Net Mode the cursor stay at lt gt Static with Soft2 Save pressed the screen shows Saved and then shows the current net mode RLS o nap o 9 Press RLS or Soft3 Quit thrice return to main interface and at this time the phone is trying to change to Static mode Press C3 button the screen shows
5. 56 CEZ VOICE TEAMING CS aani n E E E 56 6 1 3 Network features connosce ann ee a ees 57 6 1 4 Maintenance and management ccccccccccccceccccesssssseseeeeecccceeeeeeeeeauaaasseseeeeeeeeeeees 57 63 Special Teatr eSa didas 57 6 2 DIGIT CHARACTER MAP TABLE iaa 58 1 Introducing AIP 311 VoIP Phone 1 1 Thank you for your purchasing AIP 311 Thank you for your purchasing AIP 311 AIP 311 is a full feature telephone that provides voice communication over the same data network that your computer uses This phone functions not only much like a traditional phone allowing to place and receive calls and enjoy other features that traditional phone has but also 1t own many data services features which you could not expect from a traditional telephone This guide will help you easily use the various features and services available on your phone 1 2 Delivery Content Please check whether the delivery contains the following parts The base unit with display and keypad The handset The handset cable The power supply The Ethernet cable The User Manual you may download from our website IP Phone are designed to look like conventional phones the following photo shows a broad overview of the IP Phone 1 3 Keypad Key Key name Function Description Navigation key assist users for operating In idle state they have special function Left Checking Incoming call Navigation Up Checking Missed Call Right Checking lin
6. KEY Phonebook Table Phone Book Field name Default Shows the detail of current phonebook Name Shows the name corresponding to the phone number Number Shows the phone number Ring Type Shows the ring type of the incoming call Click Modify to change the selected information and click the Delete to delete the selected record Notice the maximum capability of the phonebook is 500 items 4 3 4 5 Function Key PHONE MA MN MR TN FUNCTION KEY Interface Configuration Line Key Setting SubType a peee 3 APPLY 42 Function Key Field name explanation Contrast Set contrast of screen Luminance Set luminance of screen Line Key Setting Line select SIP1 SIP2 SIP3 Dial peer or IAX2 in function key type After you set it you pick up handset or hands free press this function key then you can use the corresponding IP line a SIP1 ha Pa aa gt Memory key Set the memory key s serial number Type Memory Key settings can be stored in key storage for each number the standby or off hook select the function keys on the keyboard can call this number Line set the dial mode SIP1 SIP2 SIP3 Dialpeer IAX2 Key Key Event functions monitor state DTMF In the call send DTMF Value Set the type parameter values Line Choose which lines to use this feature Subtype Select the function parameters KeyEvent NOTICE O memory keys can be con
7. OK again to send SMS When user has new message the phone will ring there is a coin on the screen Press softkey 1 SMS select inbox use up down key and then press softkey2 OK When a number of text messages users can use up down key and press softkey2 Enter to select one to view Press softkey2 Reply and input message content finally press Soft2 Send again to reply this message The phone can also send messages by phonebook Note while user browses the message numbers new messages will be marked by new when user edits message press key that to switch input method e g ABC uppercase English input abc lowercase English input 123 digit input Korean Korean input 1f your phone s firmware version supports Korean PY if your phone s firmware version supports Chinese Oo SpeedDial function User can pre define numbers in these keys numeric key 0 9 Hook off press the defined numeric key and then input Your pre defined numbers will send out Press softkey2 SDial to set speed dial in standby a total of 12 numbers users can select by memory key Users can delete and press key that to switch input method Note 1 First 9 numbers corresponding digit key 1 9 10 number corresponding digit key 0 2 The first 10 set of numbers in standby mode press the corresponding number key and then press softkey3 Dial or Speaker key to exhale but the first 11 groups and 12 group numbers 14 without the corresp
8. Static the screen shows the IP address and gateway which were set just now if the phone could display the right time it shows that Static IP mode takes effect Setting DHCP mode 1 Press then press A twice chooses Network Then press OK or Soft2 Enter the screen will show WAN 2 Press OK or Soft2 Enter to show Net Mode Select DHCP Press Soft2 Save with 9 saved shown screen will jump to show the current net mode 3 Press or Soft3 Quit thrice back to main interface and at this time phone is trying to change to DHCP mode Press CS until the phone shows DHCP If the screen shows the IP address and gateway which were set just now 1t shows that DHCP mode takes effect 10 3 AIP 311 s basic operation 3 1 Answer calls AIP 311 will ring to indicate you when there is call incoming below is ways to answer call O Answer with hook off Take handset you can talk directly You can just hang up to finish talk O Answer with hands free Press Speaker to begin talking Press Speaker again to finish talk e Answer with headset Press Headset to answer the call press Headset again to finish talk O Using hands free instead of handset during a talk Press Speaker and hook on the handset when you use handset to speak and want to change to use hands free to speak Press Speaker again to finish talk O Using handset instead of hands free during a talk Hook of
9. VOIP PHONE Line Info In the standby screen showing the registration number of lines when the time is displayed as NULL is not registered 4 3 2 Network 4 3 2 1 WAN Config NETWORK wan MEA Beers ME dl Meteo E WAN Status o Meros OO APPLY WAN Config Field Name explanation WAN Status Current Netmask 255 255 255 0 Current Gateway 192 168 1 1 MAC Address 02 03 04 05 07 2a Active IP The current IP address of the phone Current Netmask The current Netmask address MAC Address The current MAC address of the phone Current Gateway The current Gateway IP address Get MAC Time Shows the time of getting MAC address WAN Setting Static Y DHCP PPPOE apie Please select the proper network mode according to the network condition AIP 311 provide three different network settings Oo Static If your ISP server provides you the static IP address please select this mode then finish Static Mode setting If you don t know about parameters of Static Mode setting please ask your ISP for them e DHCP In this mode you will get the information from the DHCP server automatically need not to input this information artificially o PPPoE In this mode your must input your ADSL account and password You can also refer to 3 2 1 Network setting to speed setting your network Obtain DNS server Select it to use DHCP mode to get DNS address if you don t select automatically it you will use static DNS serve
10. ask your ISP for them 18 DHCP In this mode you will get the information from the DHCP server automatically need not to input this information artificially o PPPoE In this mode your must input your ADSL account and password You can also refer to 3 2 1 Network setting to speed setting your network Choose Static IP MODE click NEXT can config the network and SIP default SIP1 simply also can browse too Click BACK can return to the last page Static IP Set Static IP Address 202 960 134 133 Alter DNS t Static IP Address Input the IP address distributed to you Netmask Input the Netmask distributed to you Gateway Input the Gateway address distributed to you DNS Domain Set DNS domain postfix When the domain which you input can not be parsed phone will automatically add this domain to the end of the domain which you input before and parse it again Primary DNS Input your primary DNS server address Alter DNS Input your standby DNS server address SIMPLE SIP SET pispiay name OOO O i Enable Register Display Name Set the display name Server Address Input your SIP server address Server Port Set your SIP server port User Name Input your SIP register account name Password Input your SIP register password Phone Number Input the phone number assigned by your VOIP service provider Enable Register Start to register or not by selecting it or not Connect Mode Static 192 168 1 179 Gateway 192
11. client to get IP in running status and network ID is also same as LAN s system will refuse to accept the IP to configure WAN So WAN s active IP will be 0 0 0 0 DD 4 3 2 2 LAN Config NETWORK f wan BEEN cos service porr once server snte When LAN IP or Bridge Mode changes the system will reboot automatically LAN Config Field name explanation LAN IP Specify LAN static IP Netmask Specify LAN Netmask Select the DHCP server of LAN port or not After you modify the DHCP Service LAN IP address phone will amend and adjust the DHCP Lease Table and save the result amended automatically according to the IP address and Netmask You need restart the phone and the DHCP server setting will take effect NAT Select NAT or not Select Bridge Mode or not If you select Bridge Mode the phone Bridge Mode 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 Notice If you choose the bridge mode the LAN configuration will be disabled 4 3 2 3 Qos Config The VOIP phone support 802 1Q P protocol and DiffServ configuration VLAN functionality can use different VLAN IDs by setting signal voice VLAN and data VLAN The VLAN application of this phone is very flexible Do not use LAN After Switchboard recelved the Broadcast Frame transmit to every other port except the send port Switchboard Broadcast 8 Lise VLAN E
12. different demands We can provide the different security level protection in terms of the different 50 resources by building a DMZ region which can provide the network level protection for the equipments environment reduce the risk which 1s caused by providing service to distrust customer and 1s the best position to put public information The following chart describes the network access control of DMZ Inner Network area SECURITY Aaa rreewace ar ven O Protocol Set FTP ALG El perp ate APPLY _ NAT Table rev ide Pon O rape pM Confia NAT Configuration Field name explanation IPSec ALG It is an encryption technology Select 1t to enable IPSec ALG the default is enable FTP is a service of connection layer which can transform intranet IP FTP ALG into extranet IP when intranet IP is sending out packet Select it to enable FTP ALG the default is enable PPTP ALG Select it enable PPTP ALG the default is enable e Shows the NAT TCP mapping table Shows the NAT UDP mapping table NAT Table Option outside port ooo 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 t
13. encryption Set sip port of each line Set ring type of each line Set hot line number of each line Configure conference number in server conference For the phone supports the transfer of certain special features server set interval time between sending bye and hanging up after the phone transfers a call Enable the option the phone will receive the notify from the server 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 30 NAT Keep Alive Enable Via rport Enable PRACK Long Contact Enable URI Convert Dial Without Register Ban Anonymous Call Enable DNS SRV Forward Type Forward Phone Number Server Type DTMF Mode RFC Protocol Edition Transport Protocol RFC Privacy Edition Subscribe Expire Time Enable Conference number MWI Number Click to Talk authorization work if it is enable 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 Disable system to support RFC3581 Via rport is special way to realize SIP NAT Enable or disable SIP PRACK function suggest use the default config Set more parameters in contact field
14. info debug level only can be displayed on telnet MAINTENANCE L AENA SYSLOG MA Bs E E Syslog Set Syslog Configuration Field name explanation Server IP Set Syslog server IP address Server Port Set Syslog server port MGR Log Level Set the level of MGR log SIP Log Level Set the level of SIP log IAX2 Log Level Set the level of IAX2 log Enable Syslog Select 1t or not to enable or disable syslog 4 3 5 3 Config Setting MAINTENANCE EAS EE conric MN al E Save Configuration Press the Save button to save the configuration files Backup Config Save all Network and VoIP settings Right Click here to Save as Config File txt Clear Configuration Press the Clear button to Clear the configuration files clear Config Setting Field name explanation you can save all changes of configurations Click the Save button Save Config all changes of configuration will be saved and be effective immediately Backup Config Right clicks on Right click here and select Save Target As 45 then you will save the config file in txt format User can restore factory default configuration and reboot the phone If you login as Admin the phone will reset all configurations and Clear Config restore factory default 1f you login as Guest the phone will reset all configurations except for VoIP accounts SIP1 2 and IAX2 and version number 4 3 5 4 Update You can u
15. n pon mode alias Suffix _ oelLength aroma ooo 060 sie aao fosu fo 135 9p000000 Joooo s060 s faaa o fno sutt o 1 x Match any single digit that is dialed If user makes the above configuration after user dials 11 digit numbers started with 13 the phone will send out O plus the dialed numbers automatically 2 Specifies a range that will match digit It may be a range a list of ranges separated by commas or a list of digits If user makes the above configuration after user dials 11 digit numbers started with from 135 to 139 the phone will send out O plus the dialed numbers automatically Use this phone you can realize dialing out via different lines without switch in web interface 34 Field name Phone number Destination Port Alias Dial Peer Option Delete Modify 156 DIAL PEER explanation There are two types of matching conditions one is full matching the other is prefix matching In the Full matching you need input your desired phone number in this blank and then you need dial the phone number to realize calling to what the phone number is mapped In the prefix matching you need input your desired prefix number and T then dial the prefix and a phone number to realize calling to what yor prefix number is mapped The prefix number supports at most 30 digits Set Destination address This is optional config item If you want to set peer to peer call please input destinat
16. 168 1 1 i Account User Name 2113 19 Display detailed information that you manual config Choose DHCP MODE click NEXT Jcan config SIP default SIP1 simply also can browse too Click BACK can return to the last page Like Static IP MODE Choose PPPoE MODE click NEXT can config the PPPoE account password and SIP default SIP1 simply also can browse too Click BACK can return to the last page Like Static IP MODE PPPOE Set PPPoE Server It will be provided by ISP Username Input your ADSL account Password Input your ADSL password Notice Click Finish button after finished your setting IP Phone will save the setting automatically and reboot After reboot you can dial by the SIP account 4 3 1 3 Call Log You can query all the outgoing through this page BASIC BEA ZO ca Loc ites Call information Called Number Call Log Field name explanation Start Time Display the start time of the outgoing record Last Time Display the conversation time of the outgoing record Called Number Display the account protocol line of the outgoing record 4 3 1 4 MMI SET BASIC ESAS MZ MOTERO MMI SET Language Selection Greeting Message Set Version VOIP PHONE V1 7 343 140 20 MMI SET Field name explanation Language Set Set the language of phone English is default The greeting message will display on LCD when phone is idle It Text Message can support 16 chars The default chars are
17. 5 0 means point to a network which network ID is C type Set the destination address mask For example 255 255 255 255 Des Mask means just point to one host 255 255 255 0 means point to a network which network ID is C type Click the Add button if you want to add a new output rule Firewall Output Rule Table 0 deny ICMP 192 168 1 14 255 255 255 0 192 168 1 118 255 255 255 0 morethan 1 Then enable out access and click the Apply button So when devices execute to ping 192 168 1 118 system will deny the request to send icmp request to 192 168 1 118 for the out access rule But if devices ping other devices which network ID is 192 168 1 0 t will be normal Rule Delete Input Output Input a Index To Be Deleted Po Delete Click the Delete button to delete the selected rule 4 3 6 3 NAT Config NAT is abbreviated from Net Address Translation it s a protocol responsible for IP address translation In other word it is responsible for transforming IP and port of private network to public also is the IP address mapping which we usually say Legal IP address UOT e SURI NAT Equipment Inner network 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 equipments not open to extranet by the corresponding isolation method according to
18. 7 4 3 CONFIGURATION VIA WEB taa aida 17 4 We BASSO a o 17 ES A A A A O N 17 AN A A A aO 18 AI CAME aos 20 ri LEY A EA RG had DA Po RO 20 AI INCUW OT ves AA A AO 21 ALAN CON is 21 AL Ze AN COMO uo 23 EAST O TA E OnE OA A et ce ae nese eee eR eee nee Re eee ne ee eae 23 E Service O REL 25 Liza DHCP SER VER ia sd db A Add a sde 26 EA nn a A 28 7 R a A TT 29 ASA E E A 29 ARI UF OATI T AAE E A T A E E E E T 32 A EA A ee rae 33 ASIA DIAL PEE RSE Mn SS ASA RE 34 ASADO A A A A A A oa 37 AA DS PCIA E E E 37 ER AR Gir he 2 ay ee mere EO ER OES On OES ORO TOON TRON NT PORE ATE ern TOOT eee ee 38 4 3 4 3 Digital Map Configuration 200 000 0000 cccccccccccccccccceeecceeaessseeeeccccceeeeeeeeaaasaeeseeeeeeeeeeeeeeeeeaaaaneasees 40 Ad PROD diia e dela 42 EIA UCA em eee 42 5 S5 A A A ee O A 44 AO AO PLOVISION ai AA AA AA ARA gaa aa a 44 AAA A O 44 ADD COnN Se 1 1 O OP e staat nav EATA EE SINN EAS 45 ree SPT A IR oC SEO O RT E 46 AI Fes ACCOUN COMO a A AAA eC Pe 47 ERR DOOL APP REN ES 47 4 E o E A o A O OTE 48 EIL MMI a e OO O E E CE CO O aba ae 48 ASO sl A REO 48 A ach asta ges dea Sec ease tect sense ab wane see ans vorenaecee esata aceetcnes 50 ADO VEN CODEN 2 sess A AE A E A A daa 52 A A A E EE T 54 S CONFIGURATION VIA KEYPA Diusapan ien a a E EEN 55 5 1 Keypad introduction sao 55 32 Cno Ei a at 55 AP PEIN UX cn ra a eee a oa eas Erea Oa Naa aa NE T tccee 56 GT SPEC INEC ATION osion is el ates 56 A II EEE EEES
19. AIP 311 VOIP PHONE User manual AIMMER TECHNOLOGY CO LIMITED www aimmertech com FA X 0086 755 83464309 TEL 0086 755 23934589 Table of Content 1 INTRODUCING AIP 311 VOIP PHONE iii dai 4 1 1 THANK YOU FOR YOUR PURCHASING AIP 311 o oooooononoccccccconononnnnnnnnnonononoccnnccnnnnnnnnnnononnnonos 4 E2DELIVERY CONITN e 4 EI RENA D EEEN P E ed N A dana enla 4 1 4 PORT FOR CONNECTING 000 A A ONTT E AE 6 2 INITIAL CONNECTING AND SETTING ices 0 as 7 Zl CONNECT THE PHONES ao 7 2 1 1 Connect to Network sui i 7 2 1 2 Power adaptor connection ccccccccccccccceecceeassssseeseeececccceeeeeeesaaaaasseeeeeeeeeeeeeeeeeeeaaas 8 22 ASIC INITIALIZATION iia task undentancast OEE NAE E AAEE TO ENAR EOE 8 2 2 1 Network Settings A ETA E E E RS 8 JAIP SBASICOPERA VION EA AAA ai 11 Debs ANS WERCALES rai 11 2 PLACE CA LES daa alada 11 A A A Dovoeatle adie 12 DEC AUG ARAN ER alan 12 Is CALL HOLD A AS A A a IN 13 3 053 WAY CONFERENCE CALD san 13 Del CAER RECORDS a SA A A A Ei io 13 DO OPE CIAL KEY S susi uc 14 Po Gy a PICKUP das 15 SLU FOUN CALLE ara aaa aida 15 311 REDIAL UN REDIA E ada 16 AZ OLIEK TODA a isa aletas 16 4 WEB CONFIGURATION cuna raices 17 4 1 INTRODUCTION OF CONFIGURATION oooooccccncnnnccooocononnnnnnnnnnnonnnnnnnononnnnnnnnnrnnnnnnnnnnnnnnnnanccnnnnnnnnns 17 ALL Ways 10 CONIL aos 17 4 1 2 Password Configuration a aaa 17 4 2 SETTING VIA WEB BROWSER unica snd couse ub A EE SE a ENEN 1
20. D VLAN ID Check Enable check if VLAN ID of a data package is not the same with the phone or a data package do not have VLAN ID the data package will be discarded 24 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 Voice Data VLAN differentiated means after using VLAN VolP signal and voice differentiated packets will add voice VLAN ID and other 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 VoIP data such as http telnet ping etc will use this value to set VLAN package NOTICE 1 Startup VLAN if set Voice Data VLAN differentiated as Undifferentiated all packets will use the Voice VLAN ID as the tag 2 Startup VLAN if set Voice Data VLAN differentiated as tag differentiated and disables the DiffServ then system will not distinguish the voice and data
21. Port 33 Set STUN Effective Time If NAT server finds that a NAT STUN Effect Time mapping is idle after time out it will release the mapping and the system need send a STUN packet to keep the mapping effective and alive Local SIP Port Set the SIP port Set Sip Line Enable Stun Choose line to set info about SIP There are 3 lines to choose You can switch by Load button Use Stun Enable Disable SIP STUN Notice SIP STUN is used to realize SIP penetration to NAT If your phone configures STUN Server IP and Port default 1s 3478 and enable SIP Stun you can use the ordinary SIP Server to realize penetration to NAT 4 3 3 4 DIAL PEER setting This functionality offers you more flexible dial rule you can refer to the following content to know how to use this dial rule When you want to dial an IP address the entry of IP addresses is very cumbersome but by this functionality you can set number 156 to replace 192 168 1 119 here When you want to dial a long distance call to Beijing you need dial an area code 010 before local phone number but you can also dial number 1 instead of 010 after we make a setting according to this dial rule For example you want to dial 01062213123 but you need dial only 162213123 to realize your long distance call after you make this setting 1T SIP 1 0 0 0 0 5060 rep 010 no suffix To save the memory and avoid abundant input of user add the follow functions Number 2 Jpestimati
22. all packets will use the Voice VLAN ID as the tag 3 Startup VLAN if set Voice Data VLAN differentiated as tag differentiated and enables the DiffServ then system will distinguish the voice and data and add the VLAN ID each other 4 Startup VLAN if set Voice Data VLAN differentiated as data untagged then the packet of the signal voice will use the Voice VLAN ID as the tag but the data packets will not take the VLAN tag 5 If Disable the VLAN regardless to set the Voice 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 it the phone will match the VLAN ID strictly When others VLAN ID not match with us the packets will discard Contrarily the phone will accept the packets with the distinct VLAN ID 7 You must gain the IP with the Static mode when you set VLAN otherwise can t gain the IP in the VLAN and also can not dial with point to point 4 3 2 4 Service Port You can set the port of telnet HTTP RTP by this page E Field name HTTP Port Telnet Port RTP Initial Port NETWORK E wan vt Tew REY service port II Bevan APPLY If modify HTTP or Telnet port you d better set it more than 1024 then restart SERVICE PORT explanation set web browse port the default is 80 port if you want to enhance system safety you d better change
23. call pick up function means appointed prefix code After making the above configuration C can dial 1 plus B s phone number to pick up A s call User can set prefix in random in the case of no affecting current dialing rules 3 10 Join call When B is calling C A can join in the existing call by inputting an appointed prefix numbers plus B or C number if B or C also supports join call The following chart shows how to configure an appointed prefix in dial peer to have join call function E 5060 SIP repsoincall aaa 15 2 means appointed prefix code After making the above configuration A can dial 2 plus B or C number to join B and C s call User can set prefix in random in the case of no affecting current dialing rules 3 11 redial unredial If B is in busy line when A calls B A will get notice busy please hang up If A want to connect B as soon as B is in idle he can use redial function at the moment and he can dials an appointed prefix number plus B s number to realize redial function What is redial function A can t not build a call with B when B is in busy then A will subscribe B s calling mode at 60 second intervals Once B is available A will get reminder of rings to hook off while A hooks off A will call B automatically If at this time A is occupied temporarily and unwilling to contact B A also can cancel the redial function by dialing an appointed prefix plus B s numb
24. de tips but does not ring Warm line set timeout to set the time line when more than warm it Warm Line Time will automatically exhaled hotline number if configured to 0 the hook immediately exhaled hotline number Set Add Delete Black list If user does not want to answer some phone calls add these phone numbers to the Black List and these calls will be rejected x and are wildcard x means matching any single digit for example 4xxx expresses any number with prefix 4 which length is 4 will be forbidden to dialed out DOT means matching any arbitrary number digit for example 6 expresses any number with prefix 6 will be forbidden to dialed Black List out if user wants to allow a number or a series of number incoming he may add the number s to the list as the white list rule the configuration rule is number for example 123456 or 1234xx Black List Means any incoming number is forbidden except for 4119 Note End with DOT when set up the white 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 can not dial out any phone number whose prefix is 001 Limit List 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
25. e status Down Checking IP info OK Enter into the phone s menu Press RLS the phone will skip to stand by mode Note DO NOT Press RLS during the configuration process or RLS Release key l ye else phone will not save the configuration modified and return to stand by status Press this key in calling mode you can hear the other side and ute mute the other side can not hear you f Envelope LED inside if blinks remind user have new voicemail Hola Temporarily hold the active call during the talking press the O key again might release the call Please refer to 3 5 call hold Transfer Trans _ F a KD Ca EX HeadSet Line1 2 3 Memory key 1 6 Soft key 1 2 3 for more details In idle mode press this key LCD will show Do Not Disturb and this phone is set to be No disturbing mode Press this key again to cancel this function Use the key to realize blind transfer or attended transfer please refers to 3 4 call transfer for more details In the idle mode press this key LCD will show call forward After this indication disappears User can configure the forward phone no of SIP1 or SIP2 Press Soft2 ON to enabled call forward function and set it to always mode Press Softl OFF to close the function of forward Place and receive calls through an optionally connected headset Here 1s Three SIP lines user could select any one to make the call if it has been registe
26. e table Set the end IP address of the lease table the network device End IP connected to LAN port will get IP address between Start IP and End IP by DHCP Netmask Set the Netmask of the lease table Gateway Set the Gateway of the lease table Lease Time Set the Lease Time of the lease table DNS Set the default DNS server IP of the lease table Click the Add button to submit and add this lease table DHCP Lease Table Delete Lease Table Name Select name of lease table click the Delete button will delete the selected lease table from DHCP lease table Select DNS Relay the default 1s enabled Click the Apply button to DNS Relay become effective Notice 1 The size of lease table can not be larger than the quantity of C network IP address We recommend you to use the default lease table and not modify 1t 27 2 If you modifies the DHCP lease table you need save the configuration and reboot 4 3 2 6 SNTP Setting time zone and SNTP Simple Network Time Protocol server according to your location you can also manually adjust date and time in this web page NETWORK ZOO E eee ESA Mo ES sure ISNTP Time Set GMT 08 00 Beijing Chongging Hong Kong Urumqi i 60 seconds Caprty SNTP Field name explanation Server Set SNTP Server IP address Time Zone Select the Time zone according to your location Time Out Set the time out the default 1s 60 seconds 12 Hours Systems Switch the time mechani
27. er before making the redial function 3 is appointed prefix code After making the above configuration A can dial 3 plus B s phone number to make the redial function A is appointed prefix code After configuration A can dial 4 to cancel redial function User can set prefix in random in the case of no affecting current dialing rules 3 12 Click to dial When user A browses in an appointed Web page user A can click to call user B via a link this link to user B then user A s phone will ring after A hooks off the phone will dial to B 16 4 Web configuration 4 1 Introduction of configuration 4 1 1 Ways to configure AIP 311 has three different ways to different users O Use phone keypad Use web browser recommendatory way e Use telnet with CLI command 4 1 2 Password Configuration There are two levels to access to phone root level and general level User with root level can browse and set all configuration parameters while user with general level can set all configuration parameters except SIP 1 2 or IAX2 s that some parameters can not be changed such as server address and port User will has different access level with different username and password O Default user with general level username guest password guest O Default user with root level username admin password admin The default password of phone screen menu is 123 4 2 Setting via web browser When thi
28. f the handset when you use hands free and want to change to use handset Just hook on to finish talk O Using headset instead of hands free during a talk In the hands free calls press Headset key you can use the headset to call After the call press headset to hang up the call O Using hands free instead of headset during a talk In the headset call press Speaker key to enter the hands free calls press Speaker key again to end the call O Using headset instead of handset during a talk In the handset call press Headset key hang up the handset to continue using the headset call After the call press the Headset key to cut off the call O Using handset instead of headset during a talk In the headset call hook off the handset after the call just hook on to finish talk 3 2 Place calls O Using handset Hook off screen will show the current using line or you could press key L1 L3 to select after getting dialing tone you could begin to dial number After finishing it press and AIP 311 will send the number and call the number When you hear a ring back tone and screen shows the callee s number it shows that the person you called is ringing If callee answers the call you can begin to talk and your phone will keep showing callee s number and counting time Just hang up to finish talk O Using headset Standby press the Headset key On screen display Enter Number Pls and hear the dialing tone you can sta
29. figuration Field name explanation End with Set Enable Disable the phone ended with FF dial Fixed Length Specify the Fixed Length of phone ending with Set the timeout of the last dial digit The call will be sent after Time out timeout Digital Rule table Below is user defined digital map rule Specifies a range that will match digit May be a range a list of ranges separated by commas or a list of digits x Match any single digit that is dialed Match any arbitrary number of digits including none Tn Indicates an additional time out period before digits are sent of n seconds in length n is mandatory and can have a value of O to 9 seconds Tn must be the last 2 characters of a dial plan If Tn is not specified it is assumed to be TO by default on all dial plans 911714 911x T4 Cause extensions 1000 8999 to be dialed immediately Cause 8 digit numbers started with 9 to be dialed immediately Cause 911 to be dialed immediately after 1t 1s entered Cause 99 to be dialed after 4 seconds Cause any number started with 9911 to be dialed 4 seconds after dialing ceases Notice End with Fixed Length Time out and Digital Map Table can be used 41 simultaneously System will stop dialing and send number according to your set rules 4 3 4 4 Phone Book You can input the name phone number and select ring type for each name here PHONE psp ff CALL SERVICE cae PHONE BOOK f Function
30. figured through the following Speed Dial function through the configuration of the key corresponding to the number of ways as shown below Fi Memory Key l4116 SIP1 Speed Dial User can press the F1 key to allocate this number by line line Push To Talk function you can press this key in standby to automatically answer the call and make each other IF 2 MemoryKey nms SIP1 PushToTalk User can be configured in accordance with push to talk function the way 4116 was the other number Then press the standby button and make it automatically answer the call 4116 O key can be configured through the following events For example Fi Key Event SIP DND 4 3 5 Maintenance 4 3 5 1 Auto Provision PHONE EN CPE ETT Pone cook EAS book Table Name number sid Phone Book Option Costete Crear Auto Provision Field name Current Config Version Server Address explanation Show the current config file s version Set FTP TFTP HTTP server IP address for auto update The address can be IP address or Domain name with subdirectory Username Set FTP server Username System will use anonymous if username keep blank Password Set FTP server Password Config File Name Set configuration file s name which need to update System will use MAC as config file name if config file name keep blank For example 000102030405 Config Encrypt Key Input the Encrypt Key if t
31. he configuration file is encrypted Protocol Type Select the Protocol type FTP TFTP or HTTP Update Interval Time Set update interval time unit is hour Different update modes 1 Disable means no update Update Mode 2 Update after reboot means update after reboot 3 Update at time interval means periodic update This option is enabled TFTP server address defaults to the value of Enable DHCP Option 66 option 66 4 3 5 2 Syslog Config 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 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 44 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 1s
32. he package to connect WAN port on the back of your phone to the Ethernet port in your workspace Since this VoIP Phone has router functionality whether you have a broadband router or not you can make direct network connect The following two figures are for your reference internet gt E ES ear ADSL Cable Arowudband Miodem Router AIP 311 ia mets E gt ADSL Cable Modem AIP 311 Shared network connection Use this method if you have a single Ethernet port in your workspace with your desktop computer already connected to it First disconnect the Ethernet cable from the computer and attach it to the WAN port on the back of your phone Next use the Ethernet cable in the package to connect LAN port on the back of your phone to your desktop computer Your IP Phone now shares a network connection with your computer The following figure is for your reference itema x ADSL Cable Modem AIP 311 Step 2 Connect the handset to the handset port by the handset cable in the package Step 3 connect the power supply plug to the AC 5V adapter port on the back of the phone Use the power cable to connect the power supply to a standard power outlet in your workspace Step 4 push the on off switch on the back of the phone to the on side then the phone s LCD screen displays Initializing wait logon Later a ready screen typically displays the date time If your LCD screen displays different informa
33. he selected mapping table DM Table 192 168 1 119 192 168 10 23 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 can not guarantee that the transmission speed reach to 100M 4 3 6 4 VPN Config This web page provides us a safe connect mode by which we can make remote access to enterprise inner network from public network That 1s to say you can set 1t to connect public networks in different areas into inner network via a special tunnel 5 Fi Be l Switchboard Rowse Physical Network RO 0 0 0 0 LI Enable VPN PN Server adar frene ame OOOO O VPN Password VPN Configuration Field name explanation VPN IP Shows the current VPN IP address YEN Mode O L2TP Cl Enable PN Select L2TP You can choose only one for current state After you select it you d better save configuration and reboot your phone Enable VPN Select 1
34. ion IP address or domain name If you want to use this dial rule on SIP2 line you need input 255 255 255 255 or 0 0 0 2 in it SIP3 into 0 0 0 3 Set the Signal port the default is 5060 for SIP Set alias This is optional config item If you don t set Alias it will show no alias Note There are four types of aliases 1 add xxx it means that you need dial xxx in front of phone number which will reduce dialing number length 2 all xxx it means that xxx will replace some phone number 3 del It means that phone will delete the number with length appointed 4 Rep It means that phone will replace the number with length and number appointed You can refer to the following examples of different alias application to know more how to use different aliases and this dial rule Call Mode Suffix Select different signal protocol SIP or AX2 Set suffix this is optional config item It will show no suffix if you 35 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 Introduction of how to set up dial peer to implement switch between multi SIP lines Del Length 1 9T mapping If you have registered a SIP1 server and set dial peer according to the above table all calls will be sen
35. it into non 80 standard port Example The IP address is 192 168 1 70 and the port value is 8090 the accessing address is http 192 168 1 70 8090 Set Telnet Port the default is 23 You can change the value into others Example The IP address is 192 168 1 70 the telnet port value is 8023 the accessing address 1s telnet 192 168 1 70 8023 Set the RTP Initial Port Itis 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 4 3 2 5 DHCP SERVER ee NETWORK m MON DIOE KLE MESA OHC server BEN 222 pendan oae fomes lan 192 168 10 1 192 168 10 30 1440 ooo e DHCP Lease Table Delete o a E DNS relay Setting DHCP SERVER Field name explanation 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 DHCP Lease Table lan 192 168 10 1 192 168 10 30 14400 255 255 255 0 192 168 10 1 192 168 10 1 Shows the DHCP Lease Table the unit of Lease time is Minute Lease Table Name Specify the name of the lease table Start IP Set the start IP address of the leas
36. key or else the phone will not save the configuration and will return to standby interface Setting PPPoE mode for ADSL connection 1 Get PPPoE account and password first 2 Press and press ES twice screen will show Network Then press Soft2 Enter or OK the LCD screen will display WAN 3 Press Soft2 Enter then press choose PPPoE 4 Press Softl Edit the screen will display Account The screen will show the current account information Press Softl Del to delete it then input your PPPoE account and press Soft2 Save With saved displayed screen will jump to password settings 5 Press Soft2 Edit again then input your PPPoE password and press Soft2 OK With Saved 8 displayed screen will display the current password and confirm you need input the password again after confirm press soft2 OK to save the Account and password 6 Press Soft3 Quit once return to Net Mode Press Soft2 Save the screen will show Saved and then jump to show the current net mode 7 Press or Soft3 Quit thrice return to standby at this time the phone is trying to change to PPPoE mode Press OR for checking the status If the screen shows Negotiating 1t shows that the phone is trying to access to the PPPoE Server 1f 1t shows an IP address then the phone has already get IP with PPPoE Setting Static IP mode static ADSL Cable or no PPPoE DHCP network
37. ling line 1f one line can t be connected the phone can automatically switch to other line to call DTMF Relay support SIP info DTMF Relay RFC2833 SIP application SIP Call forward transfer blind attended hold waiting 3 way talking sms pickup joincall redial unredial multi line Call control features Flexible dial map hotline empty calling No reject service black list for reject authenticated call limit call no disturb caller ID Flexible deer peer rule Support phonebook 500 records Incoming calls outgoing calls missing calls Each supports 100 records Support IAX2 Phonebook supports vcard standard 12 24 hours time display Support daylight saving time Support path gruu Support SIP Privacy Support SMS Support WMI Support Speed dial Support XML 6 1 3 Network features WAN LAN support bridge and router model Support PPPoE for xDSL Support basic NAT and NAPT Support VLAN optional voice vlan data vlan NAT Penetrate Stun Penetrate Support DMZ Support VPN L2TP function Wan Port supports main DNS and secondary DNS server can select dynamically to get DNS in DHCP mode or statically set DNS address Support DHCP client on WAN Support DHCP server on LAN QoS with DiffServ Network tools in telnet server including ping trace route telnet client 6 1 4 Maintenance and management Upgrade firmware through POST mode Web telnet and keypad management Management with different account righ
38. me device own IP which is pre specified access to the MMI of the phone SECURITY MMI FILTER MMI Filter Table MMI Filter Table Set MMI Filter Table Set O MMI Filter to config and manage the phone Field name explanation MMI filter Table Start IP End IP pti 192 168 1 1 I 192 168 1 20 Modify MMI Filter IP Table list MMI Filter Table Set Add or delete the IP address segments that access to the phone Set initial IP address in the Start IP column Set end IP address in the End IP column and click Add to add this IP segment You can also click Delete to delete the selected IP segment MMI Filter Select it or not to enable or disable MMI Filter Click Apply to make it effective Notice Do not set your visiting IP outside the MMI filter range otherwise you can not logon through the web 4 3 6 2 Firewall 48 SECURITY FIREWALL Firewall Type O In_access Enable coi hie Sa Firewall Input Rule Table Firewall Output Rule Table Indexjpeny Permilprotocollsre addr src Mask Des Addr pes Mask Range Port Firewall Set Input outpat input reader o pit i Protocol Type CIO CIN EEN po morethav besmask Rule Delete Firewall Configuration In this web interface you can set up firewall to prevent unauthorized Internet users from accessing private networks connected to the Internet input rule or prevent unauthorized private network devices from accessing the Internet ou
39. ne s power cut or reboot call records will be discarded O Missed call and screen displays Missed Call with the number and time of missed call User can also use ML Se to browse the missed call records or press Softl Option to 13 check the details of this record then press Soft2 EDial again to change the current number Pressing Soft2 Dial will call this number directly if user don t modify the number If there is no missed call screen will show List Is Empty O Incoming call Press ha and screen displays Incoming Call by pressing Con a to browse the records or press Softl Option to check the details of this record then press Soft2 EDial again to change the current number Pressing Soft2 Dial will call this number directly if user don t modify the number If there is no incoming call screen will show List Is Empty Oo Dialed call Sd d to browse the dialed call records or press Softl Option to check the details of this record then press Soft2 EDial again to change the current number and use Pressing Soft2 Dial will call this number directly if user don t modify the number If there is no dialed call screen will show List Is Empty 3 8 Special keys Oo SMS function In the standby press Softkey1 SMS then press Softl new key After inputting SMS content press Soft2 send key to input callee s number next press Soft2
40. number in stand by mode but first user need to add and edit SDial no By pressing Soft2 SDial to edit and save the number to be a SDial number In this way user could make a call only press the number and Soft3 Dial eo Miultiple way call If user has 2 line calls and wants to invite the third party during the call they can press Softl Conf or Soft2 Transf New CALL press Softl OK enter the number then press Soft2 Send and wait for the other party to answer When the multiple way calls you can press the arrow keys to select a call 3 3 End calls O Hang up with handset hook on Hook on to finish talking O Hang up with hands free Press Speaker key to finish talk when phone is in hands free status O Hang up with headset If you are in the headset call press Headset key to end the call O Hang up an active call with 2 calls When there are two calls user might use Softl Switch to switch to the call you want to hang up first Then press Soft3 Close to finish talk and phone will switch to the other call automatically 3 4 Call transfer e Blind Transfer 12 During talk press or Soft2 Transf and then dial the number that you want to transfer to and finished by Phone will transfer the current call to the third party After finishing transfer the call you talk to will be hanged up e Attended Transfer E Trans l During talk press Trans or Soft2 Transf then input the number that y
41. o test number If IAX2 server supports echo test and echo test number is non numeric system could set an echo test number to 539 replace the echo test text So user can dial the numeric number to test echo voice test This function is provided with server to make endpoint to test whether endpoint could talk through server normally Echo Test Text Specify echo test text s name Refresh Time Set expire time of A X2 server register you can set it between 60 and 3600 seconds Enable Register Start to register the AX2 server or not by selecting it or not Enable G 729 Enable or disable code G 729 by selecting it or not 4 3 3 3 Stun Config In this web page you can config SIP STUN STUN By STUN server the phone in private network could know the type of NAT and the NAT mapping IP and port of SIP The phone might register itself to SIP server with global IP and port to realize the device both calling and being called in private network Send request to Stun server AT from 5060 port apping port 12345 Stun server tell ant to receive data from S060 Public Network Phe Gateway STUN Server Set Sip Line Enable STUN siiv toa Usestun O STUN Field name explanation STUN NAT Transverse Shows STUN NAT Transverse estimation true means STUN can penetrate NAT while False means not STUN Server Addr Set your SIP STUN Server IP address STUN Server Port Set your SIP STUN Server
42. ode Enable Session Timer Answer With Single Codec Auto TCP Enable Strict Proxy Enable GRUU Enable Display name Quote 4 3 3 2 IAX2 Config Field name Register Status IAX2 Server Addr TAX2 Server Port Account Name Account Password Phone Number Local Port Voice Mail Number Voice Mail Text Echo Test Number Enable Disable Signal Encrypt Enable Disable RTP Encrypt Set Enable Disable Session Timer whether support RFC4028 It will refresh the SIP sessions Enable Disable the function when call is incoming phone replies SIP message with just one codec which phone supports Set to use automatically TCP protocol to guarantee usability of transport as message is above 1300 byte Support the special SIP server when phone receives the packets sent from server phone will use the source IP address not the address in via field Set to support GRUU Set to make quotation mark to display name as the phone sends out signal in order to be compatible with server IA X2 Config explanation Shows if the phone has been registered the AX2 server or not Input your AX2 server address Set your A X2 server port the default is 4569 Input your IAX2 register account name Input your IAX2 register password Input your assigned phone number usually it is same you re your IAX2 account name Set your local sport the default is 4569 Specify the voice mail s number Specify the voice mail s name Set ech
43. onding number key is required to enter SDial menu to find the set of numbers by Corresponds memory key or softkey3 Dial button to exhaled O Realize Secondary Dial by Dialing for only one time When you make secondary dial in off hook hands free or standby pre input mode press hold button to postpone input and screen display will show One stands for 2 seconds For example you input 12345 the phone will send DTMF 45 2 seconds after the phone call 123 12344445 will make phone send DTMF 45 at 6 seconds interval O Message waiting indication After you set it you can pick up or hands free then press E to listen to record in server when you have new voice message O Phone book search function In the Chinese version users can be retrieved by the corresponding initials Chinese name which simplifies the steps in the phone book to find contacts For example contact name is Zhang San contact number is 123 When you enter the phonebook you can press 9 key to select letter z all the numbers of beginning with z will display on the phone you can select the one you want to search by press up down key 3 9 Call pickup Call pickup is implemented by simulating pickup function of PBX it s that when A calls B B rings but no answer at this moment C can hook off and input an appointed prefix plus B s number pick up A s call and talk with A The following chart shows how to configure an appointed prefix in dial peer to have
44. ou want to transfer to T and press Soft2 Send After that third party answers then press LS complete the transfer You need enable call waiting and call transfer first If there are two calls you can just talk to one and keep hold to the other one The one who is keep hold can not speak to you or hear from you eo Alert Transfer T During the talk press or Soft2 Transf firstly then press Soft2 Send after inputting the e Trans number that you want to transfer You are waiting for connection now press Trans or Soft2 Transf and the transfer will be done To use this feature you need enable call waiting and call transfer first 3 5 Call hold _ Hold During talking user could press co to hold the current call Press again to return the call or switch the call active 3 6 3 way conference call User can press Softl Conf to dial the line2 press Soft Answer to answer the call directly if this call is from line2 during talking with linel After line2 connect user can press Soft2 Conf select another way into the three way calling number then press softkey1 OK to enter into conference mode To back to linel from conference please press Soft1 Split to end the call RLS please press Soft3 Close or me RS 3 7 Call records AIP 311 supports 100 items of missed call 100 items of incoming call and 100 items of dialed call If the records are full the newest will replace the oldest If pho
45. pdate your configuration with your config file in this web page MAINTENANCE A ct Mee ccm roar MT Masser Web Update Field name explanation Click the browse button find out the config file saved before or Web Update provided by manufacturer download it to the phone directly press Update to save You can also update downloaded update file logo picture ring mmiset file by web Server Set the FTP TFTP server address for download upload The address can be IP address or Domain name with subdirectory Username Set the FTP server Username for download upload Password Set the FTP server password for download upload File name Set the name of update file or config file The default name is the MAC of the phone such as 000102030405 Notice You can modify the exported config file And you can also download config file which includes several modules that need to be imported For example you can download a config file just keep with SIP module After reboot other modules of system still use previous setting and are not lost Action type that system want to execute 1 Application update download system update file Type 2 Config file export Upload the config file to FTP TFTP server name and save it 3 Config fie import Download the config file to phone from FTP TFTP server The configuration will be effective after the phone is reset 4 Phone book export vcf Upload the phonebook file to FTP TFTP server name and sa
46. poe ett Call Service Specify Hotline number If you set the number you can not dial Hotline any other numbers 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 P2P IP Prefix you dial only 119 to reach 192 168 1 119 Default is If there is no Set it means to disable dialing IP If select it the phone will auto answer when there is an incoming Auto Answer i call Select NO Disturb the phone will reject any incoming call the Do Not Disturb callers will be reminded by busy but any outgoing call from the phone will work well If you select Ban Outgoing to enable it and you can not dial out Ban Outgoing oe Enable Call Transfer Enable Call Transfer by selecting it Enable Call Waiting Enable Call Waiting by selecting it Enable Three Way Enable Three Way Call Call If select it the phone will accept the call even if the called number Accept Any Call is not belong to the phone The phone will hang up and return to standby automatically at Auto Handdown hands free mode Configuration automatically hang time if it is hands free mode Auto Handdown Time then more than auto handdown time the phone automatically returns to standby mode if the handle pattern then more than auto 39 handdown time it automatically put a dial tone Configuring the mute mode if the mute mode calls Icd will flash Mute Mo
47. r The default is selecting it Static IP Address 192 168 1 1 9 0056 134 133 Alter DNS 1202 96 128 68 Auto DNS If you use static mode you need set it IP Address Input the IP address distributed to you Netmask Input the Netmask distributed to you Gateway Input the Gateway address distributed to you Set DNS domain postfix When the domain which you input can DNS Domain not be parsed phone will automatically add this domain to the end of the domain which you input before and parse it again Primary DNS Input your primary DNS server address Alter DNS Input your standby DNS server address PPPOE Server ANY Username userl23 Password AA If you uses PPPoE mode you need to make the above setting PPPoE Server It will be provided by ISP Username Input your ADSL account Password Input your ADSL password Notice 1 Click Apply button after finished your setting IP Phone will save the setting automatically and new setting will take effect 2 If you modify the IP address the web will not response by the old IP address Your need input new IP address in the address column to logon in the phone 3 If networks ID which is DHCP server distributed is same as network ID which is used by LAN of system system will use the DHCP IP to set WAN and modify LAN s networks ID for example system will change LAN IP from 192 168 10 1 to 192 168 11 1 when system uses DHCP client to get IP in startup if system uses DHCP
48. r dialed number if your dialed number starts with your set phone number You need set Phone Number Alias and Delete Length Phone number is XXXT and Alias is rep Xxx If your dialed phone number starts with your set phone number the first digits same as your set phone number will be replaced by the alias number specified and New phone number will be send out If your dialed phone number starts with your set phone number The phone will send out your dialed phone number adding suffix number When you dial 2 the SIP1 server will receive 33334444 When you dial 8309 the SIPI server will receive 07558309 When you dial 0106228 the SIP1 server will receive 86106228 When you dial 147 the SIP1 server will receive 1470011 4 3 4 Phone 4 3 4 1 DSP Config In this page you can configure voice codec input output volume and so on aTa PHONE ose MAME MAA MR Map da DSP Configuration First Codec g711Ulaw64k v 9711Alaw64k v Third Codec 9729 Fourth Codec 9723 we Na Codec A 32 E Codec g722 kai Seventh Codec AMR AMRPayload Type 108 96 127 a a Handfree Volume 1 9 1 9 China Y 73 rmestanpe_ so nome santa a 999 en en 96 427 DSP Configuration Field name explanation First Codec The fist preferential DSP codec G 711A u G722 G723 G 729 G 726 AMR Second Codec The second preferential DSP codec G 711A u G 722 G 723
49. red Turn down or turn up the volume by pressing these two keys 1 In the hook off hands free mode use the key to dial the last call number 2 In stand by mode it has a function to check the OUTGOING CALL Make the phone into hands free mode If the light blinking indicate the phone has missed call Users could store their commonly used number in these keys and call for them as speed dial Keys combination include functions such as SMS SDial PBook Answer Conf Enter Save Quit Edit Redial and so on 1 4 Port for connecting Port Port name description Power switch Input SV AC 1A WAN 10 100M Connect it to Network LAN 10 100M Connect it to PC Handset Port type RJ 9 connector Headset Port type RJ 9 connector Headset Port type 3 5mm jack AIP 311 provide two Ethernet ports and a power adaptor Also has two headset interfaces with RJ 9 port and 3 5mm jack Please refer to safety notes of this manual carefully before power adaptor is connected 2 Initial connecting and Setting 2 1 Connect the phone 2 1 1 Connect to network Step 1 Connect the IP Phone to the corporate IP telephony network Before you connect the phone to the network please check 1f your network can work normally You can do this in one of two ways depending on how your workspace is set up Direct network connection by this method you need at least one available Ethernet port in your workspace Use the Ethernet cable in t
50. roadcast Frame After Switchboard received the Broadcast Frarne only transmit it to other port which belang ta same VLAN with send port Broadcast Frame i Broadcast IS Es Domain Broadcast Domain In chart 1 there is a layer 2 switches without setting VLAN Any broadcast frame will be transmitted to the other ports except the send port For example a broadcast information is sent out from port 1 then transmitted to port 2 3and 4 In chart 2 red and blue indicate two different VLANs in the switch and port 1 and port 2 belong to red VLAN port 3 and port 4 belong to blue VLAN If a broadcast frame is sent out from port 1 switch will transmit it to port 2 the other port in the red VLAN and not transmit it to port3 and port 4 in blue VLAN By this means VLAN divide the broadcast domain via restricting the range of broadcast frame transmition Note chart 2 use red and blue to identify the different VLAN but in practice VLAN uses different VLAN IDs to identify NETWORK qos ESA Ieee EA l VLAN ID Check Enable Voice Data VLAN differentiated Undifferentiated v C DiffServ Enable DiffServ Value Ox b Voice 802 1P Priority 0 0 7 Data 802 1P Priority 0 0 7 Voice VLAN ID 256 0 4095 Data VLAN 1D 254 0 4095 QoS Configuration Field name explanation VLAN Enable Before select 1t to enable VLAN you need enable Bridge mode in LAN config Enable VLAN ID check by selecting it After enable VLAN I
51. rt dialing After finishing it press or press the softkey2 Send AIP 311 can immediately begin connecting with each other When you hear a ring back tone and screen shows the callee s number it shows that the person you called is ringing If callee answers the call you can begin to talk and your phone will keep showing callee s number and counting time Just press Headset key to finish talk O Using hands free Press Speaker key screen will show the current using line or you could press key L1 L3 to select after getting dialing tone you could begin to dial number After finishing it press and AIP 311 will send the number and call the number When you hear the ringback tone and screen shows the callee s number it shows that the person you called is ringing If callee answers the call you can begin to talk and your phone will keep showing callee s number and counting time Press Speaker key again to finish talk O Using directory Press Soft3 PBook in stand by mode you will access to phonebook If there are many persons records stored in the directory you can use as di Mee to select number or press the first character of the name for searching the person which you want to contact Press A to forward and press Wa to backward Press Soft2 Dial to dial the current number shown on the screen O Speed dial Speed dial means user can make calls directly without hook off or using hands free User can dial
52. s phone and PC are connected to network enter the IP address of the wan port in this phone as the URL e g http xxx xxx xxx xxx or http Kxx xxxX XXX XXX XXXX If you do not know the IP address you can look it up on the phone s display by pressin gii button The login page is as below picture Username Password 4 3 Configuration via WEB 4 3 1 BASIC 4 3 1 1 Status BASIC status RE RAMOS NES connect P aaaeeeaeo e ponce TP Address 192 168 10 1 00 01 0e 61 00 50 IP Address 192 168 1 40 Unapplied Status Field name Explanation Shows the configuration information on WAN and LAN port Network including the connect mode of WAN port Static DHCP PPPoE MAC address the IP address of WAN port and LAN port ON or OFF of DHCP mode of LAN port Phone Number Shows the phone numbers provided by the SIP LINE 1 3 servers and IA X2 The last line shows the version number and issued date 4 3 1 2 Wizard BASIC SA WIZARD MAMA MES Network Mode Select Static IP MODE DHCP MODE al PPPoE MODE Wizard Field Name Explanation Static IP MODE DHCP MODE Please select the proper network mode according to the network condition AIP 311 provide three different network settings o Static If your ISP server provides you the static IP address please select this mode then finish Static Mode setting If you don t know about parameters of Static Mode setting please
53. sm between 12 hours and 24 hours Default is 24 hours mode SNTP Select the SNTP and click Apply to make the SNTP Times effective Enable Daylight Enable daylight saving time Time shift minutes Setup the variety length Month Setup stat and end month Week Setup start and end week Day Setup start and end day Hour Setup start and end hours Minute Setup start and end minutes 28 Manual Timeset Notice You need specify the above all items 4 3 3 VOIP 4 3 3 1 SIP Config Set your SIP server in the following interface Field name VOIP ME oa stun lo peer O i SIP Line Select AA II el severe rw servers AAA Advanced Set NAT Keep Alive Interval 69 seconds Fever Pran Phone Number SS A COMMON a T ormr_arcz833 a e A Long Contat O oree OO E n a E sie ry__ t ___ BH SIP Config explanation SIP Line Select 29 Choose line to set info about SIP there are 3 lines to choose You can switch by Load button Register Status Server Name Server Address Server Port Account Name Password Phone Number Display Name Proxy Server Address Proxy Server Port Proxy Username Proxy Password Domain Realm Enable Register Register Expire Time NAT Keep Alive Interval User Agent Signal Key Media Key Local port Ring type Hot line Number Conference Number Transfer Expire Time Enable subscribe Enable Keep
54. t LCD and WEB configuration can be modified into requested language and support multi language dynamically shifted Upgrade firmware through HTTP FTP or TFTP Telnet remote management upload download setting file Support Syslog Support Auto Provisioning upgrade firmware or configuration file 6 1 5 Special features Support 3 softkeys 6 memory keys Navigation key RLS Pbook MWI HOLD Trans Mute L1 L3 Vol Redial 57 6 2 Digit character map table Keypad Keypad 58
55. t or not to enable or disable VPN L2TP VPN Server Addr Set VPN L2TP Server IP address VPN User Name Set User Name access to VPN L2TP Server VPN Password Set Password access to VPN L2TP Server ma 4 3 7 Logout System Logout Click Logout and you will exit web page If you want to enter 1t next time you need input user name and password again 54 5 Configuration via Keypad 5 1 Keypad introduction Use need input password default 123 when login the menu system config 5 2 Menu Tree Menu Tree List 6 Appendix 6 1 Specification 6 1 1 Hardware Adapter Input 100 240V Input Output Output SV 1A port 10 100Base T_ RJ 45 for LAN LAN 10 100Base T RJ 45 for PC Idle 2 5W Active 2 8W LCD Size 128x96 53 5 x 70mm Operation Temperature 0 40 C Relative Humidity 10 65 SDRAM 16MB Flash 4MB Dimension Lx Wx H 11 6X8X3 in 295 X 205 X75mm 6 1 2 Voice features SIP supports 3 SIP servers Support SIP 2 0 RFC3261 and correlative RFCs Codec G 711A u G 723 1 high low G 729a b G 722 G 726 Echo cancellation G 168 Compliance in LEC additional acoustic echo cancellation AEC can reach 96ms max filter length in hands free mode Support Voice Gain Setting VAD CNG Support full duplex hands free HD Voice SIP support SIP domain SIP authentication none basic MD5 DNS name of server Peer to Peer IP call Automatically select cal
56. t via SIP1 server when you press the numeric key 9 in front of dialing destination phone numbers 8T mapping If you have registered a Private SIP2 server and set dial peer according to the above table all calls will be sent via SIP2 server when you press the numeric key 8 in front of dialing destination phone numbers the rule of 2T means user need to dial the number with prefix 2 1f he want to dial via AX2 server Examples of different alias application Set by web explanation example You need set phone number If you dial 93333 Phone Number Destination Alias and Delete the SIP2 server will Length receive 3333 Phone number is XXXT canmodo Destination 1s 255 255 255 255 0 0 0 2 and Alias is del This means any phone No that starts with your set phone number will be sent via SIP2 line after the first several digits of your dialed phone number are deleted according to delete length 36 Phone Number Alias optional Delete Length opio Prone numer pontono optio lode Suffix optional Delete Length optional Alias il Call M Destination optional Miasto ional e optional D Destination optional ional elete Length optional This setting will realize speed dial function after you dialing the numeric key 2 the number after all will be sent out The phone will automatically send out alias number adding you
57. tion from the above you need refer to the next section Initial setting to set your network online mode If your VoIP phone registers into corporate IP telephony Server your phone is ready to use 2 1 2 Power adaptor connection Make sure that the power you use is comply with the parameters of power adaptor 1 Plug power adaptor to power socket 2 Plug power adaptor s AC output to the AC5V port of AIP 311 to start up 3 There will be displayed black line and initializing wait logon on the screen After finishing startup phone will show greeting current date and time and so forth 4 If phone has registered to the server you can place or answer calls 2 2 Basic Initialization AIP 311 is provided with a plenty of functions and parameters for configuration User needs some network and VoIP knowledge so that user could understand the meanings of parameters In order to make user use the phone more easily and convenient there are basic configurations introduced which is mandatory to ensure phone calls 2 2 1 Network settings Make sure that network is connected already before setting network of phone AIP 311 uses DHCP to get WAN IP configurations so phone could access to network as long as there is DHCP server in it If there is no DHCP server available phone has to be changed WAN network setting to Static IP or PPPoE RLS gt Note during setting network parameter in menu Please don t press the CS
58. to dialed out means matching any arbitrary number digit For example 6 expresses any number with prefix 6 will be forbidden to dialed out Notice Black List and Limit List can record at most10 items respectively 4 3 4 3 Digital Map Configuration This system supports 4 dial modes 1 End with dial your desired number and then press 2 Fixed Length the phone will intersect the number according to your specified length 3 Time Out After you stop dialing and waiting time out system will send the number collected 4 User defined you can customize digital map rules to make dialing more flexible It is realized by defining the prefix of phone number and number length of dialing In order to keep some users secondary dialing manner when dialing the external line with PBX 40 phone can be added a special rule to realize it so user can dial a number as external line prefix and get the secondary dial tone to keep dial the external number After finishing dialing phone will send the prefix and external number totally to the server For example there is a rule 9 xxxxxxxx in the digital map table After dialing 9 phone will send the secondary dial tone user may keep going dialing After finished phone will call the number which starts with 9 actually the number sent out is 9 digit with 9 PHONE E E MEAR ccm map LEO AT Digital Map Set arm e I IU Fixed Length j Time Out El Digital Map Con
59. tput rule Firewall supports two types of rules input access rule and output access rule Each type supports at most 10 items Through this web page you could set up and enable disable firewall with input output rules System could prevent unauthorized access or access other networks set in rules for security Firewall is also called access list is a simple implementation of a Cisco like access list firewall It supports two access lists one for filtering input packets and the other for filtering output packets Each kind of list could be added 10 items We will give you an instance for your reference C In_access Enable C Out_access Enable Field name explanation In access enable Select it to Enable in_ access rule out access enable Select it to Enable out_ access rule 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 Des Addr Set the destination address It can be IP address network address complete address 0 0 0 0 or network address similar to Set the source address mask For example 255 255 255 255 means 49 Src Mask just point to one host 255 255 25
60. ve it 5 PhoneBook import vcf Download the phonebook file to phone 46 from FTP TFTP server Protocol Select FTP TFTP server 4 3 5 5 Account Config You can add or delete user account and change the authority of each user account in this web page MAINTENANCE SA METI ol MS ACCOUNT Eae Set Keyboard Password User Level Root General Account Option dni g Account Configuration Field name explanation Keyboard Password Set the password for entering the setting menu of the phone by the phone s key board The password is digit User Set guest General This table shows the current user existed User Name Set account user name User Level Set user level Root user has the right to modify configuration General can only read Password Set the password Confirm Confirm the password Select the account and click the Modify to modify the selected account and click the Delete to delete the selected account General user only can add the user whose level is General 4 3 5 6 Reboot 47 MAINTENANCE SA Mc Ma Mi Ms a REBOOT Reboot Phone Press the Reboot button to reboot Phone If you modified some configurations which need the phone s reboot to be effective you need click the Reboot then the phone will reboot immediately Notice Before reboot you need confirm that you have saved all configurations 4 3 6 Security 4 3 6 1 MMI Filter MMI Filter User could make so

Download Pdf Manuals

Related Search

Related Contents

Telemecanique ASISAFEMON1 /..1B ASISAFEMON2 /..2B  取扱説明書 - 日立工機  Réservez votre expo - Espace des sciences  Philips DS1155  1. Seguridad 2. Información general del producto 3  Manual del Producto (CONTROL  Manual de Instalación y Mantenimiento.  KitchenAid KESK901SBL06 User's Manual  JVC KS-BTA200 User's Manual  

Copyright © All rights reserved. Failed to retrieve file