VOI-7010 / VOI-7011 SIP IP Telephone
Contents
1. Password E mail Address DDNS Server DDNS Server List Type Wild Card BACKMX On Of Off Line On Off 32 You need to have a DDNS account before configuring the DDNS setting Usually most of the VoIP applications are working with a SIP Proxy Server Nonetheless you may have a DDNS account with a public IP address and others can call you via the DDNS account Example DDNS On OOR Host Name levelone dyndns org User Name levelone Password s E mail Address DDNS Server DDNS Server List members dyndns org y Type dyndns Y Wild Card on M BACKMX OOn OOF Of Line OOn Gor In this example the other user can place VoIP calls to your IP Phone directly by your domain address 33 3 4 SIP Settings You can setup the Service Domain Port Settings Codec Settings RTP Setting RPort Setting and Other Settings for SIP Proxy Server registrations in this page SIP Setting Service Domain Codec Codec ID Other 34 Understanding the SIP SIP the Session Initiation Protocol is a signalling protocol for Internet conferencing telephony presence events notification and instant messaging SIP was developed within the IETF MMUSIC Multiparty Multimedia Session Control working group with work proceeding since September 1999 in the IETF SIP working group SIP enabled PBXes and or SIP User Agents utilize the Session Initiation Protocol2. You could enable disable the auto configuration setting in this page Auto Configuration Off C By TFTP By FTP TFTP Server foo00 FTP Server o 0 00 FTP Username N FTP Password Submit Reset Note Auto Config is idea for ITSP or large network group to deploy VoIP devices easily 46 ICMP Setting The ICMP function is to echo when someone ping this device This can prevent from hacker attacking the device by not echoing ICMP Setting You could enable disable the ICMP setting in this page ICMP Not Echo On Off Submit Reset ICMP Not Echo ICMP is used to acknowledge and echo for the Ping request IP Phone will echo for the IP Ping request at default Selecting ON for ICMP Not Echo will ignore the IP Ping request and keep silent This is sometime useful for network security 3 6 User Password You may create the login name and password in this page User Password You could change the login username password in this page New username New password Confirmed password 47 3 7 Save Change You must save the changes you have made and click the Save button After clicking the Save button the IP Phone will save the new settings into ROM and reboot it automatically Save Changes You have to save changes to effect them Save Changes 3 8 Update User can update the IP Phone firmware when new firmware is available Make sure no power off during the firmware
3. Registered in the Register Status it indicates a successful registration to the ITSP and the REG LED will On The IP Phone is then ready for VoIP call If you have more than one SIP account please follow the steps to register to other ITSPs Note After you finished the setting please click the Submit button and click Save Change 37 2833 O Inband DTMF O Send DTMF SIP Info SIP Port 5060 102465535 RTP Port 60000 1024 65535 STUN Setting STUN OOn off STUN Server stun xten com STUN Port 3478 DTMF Setting You can setup the options for DTMF function in this page The options include RFC2833 Outband DTMF Inband DTMF and Send DTMF SIP info The default is set at Inband DTMF If you are making two stage callings for extension to PSTN you may need to select Outband DTMF option Port Setting The SIP Port and RTP Port numbers are default at 5060 and 60000 respectively The RTP port number must be even number If you have more than one VoIP phones under the same NAT router it is recommended that different RTP port numbers be assigned to each of IP Phones STUN Setting The STUN function must be enabled to work properly behind NAT when registered in SIP server You may enter the STUN server IP address and the STUN port number Please check your ITSP for STUN information 39 Codec You can setup the Codec priority RTP packet length and VAD function in this page Codecs basical
4. Client embedded VOI 7100 PoE Complies with 802 3af PoE Standard 1 2 Packing Contents Open the shipping cartons of the Switch and carefully unpacks its contents The carton should contain the following items SIP IP Telephone Power Adaptor Gat 5 Cable CD User Manual If any item is found missing or damaged please contact your local reseller for replacement 2 Hardware Description 2 1 LCD Display and Keypads The LCD display and keypads of IP Phone are as the following Volume d Up Down Handsfree Number Keypads 2 2 Front Panel VOI 7000 VOI 7100 Memory Card Use the memory card as a name index for speed dialler or extensions 2 3 Connection Diagram Without Broadband Router Power WAN Internet Note Public Switched Telephone Network PSTN which refers to the international telephone system based on copper wires carrying analog voice data Telephone service carried by the PSTN is often called plain old telephone service POTS 2 4 Installation 1 Connect IP Phone RJ45 WAN port to NAT Router using a Category 5 LAN cable 2 Connect IP Phone RJ45 LAN port to Notebook PC using a Category 5 LAN cable 3 Connect DC power adaptor and the LCD panel will start showing Loading Program and System Initialized 4 The LCD panel will show Date Time and No service without SIP registration or lt phone number gt after successful SIP registr
5. forward the calls It requires Submit Save and Reboot to activate new settings 18 SNTP You can setup the primary and second SNTP Server IP Address to get the date time information You may also setthe Time Zone and how long need to synchronize again When you finished the setting please click the Submit button SNTP Settings You could set the SNTP servers in this page SNTP Primary Server Secondary Server Time Zone Sync Time On OOR time window com 208 184 49 9 GMT 08 100 hh mm fo Ho dd hh mm SNTP Simple Network Time Protocol SNTP is an acronym that stands for Simple Network Time Protocol SNTP enables IP Phone to synchronizing the clocks over Internet Time Servers which it is very precise timekeeping 19 Volume Raise or lower the sound level by using the Volume Control For example if it is difficult to hear the other party s voice raise the Handset Volume or If the other party has difficulty hearing you raise the Handset Gain level Volume Setting You could set the volume of your phone in this page Handset Volume Speaker Volume Ringer Volume Handset Gain Speaker Gain Handset Vol Speaker Vol Ringer Vol Handset Gain Speaker Gain 10 0 15 10 0 15 6 0 10 10 0 15 9 0 15 Submit Reset Set the volume to hear from the handset Set the volume to hear from the Speaker Se
6. pressing flash key while holding the current call You may switch back to previous call by pressing flash key again i Call Waiting Setting You could enable disable the call waiting setting in this page Call Waiting On OOR Note Flash key means On hook and Off hook in short period without hanging up the call 26 3 3 Network You can check the Network status and configure the WAN LAN DDNS VLAN DMZ Virtual Server and PPTP settings in this section Network Status WAN LAN DDNS VLAN DMZ Virtual Server PPTP 27 Network Status You can check and show the current Network settings in this page Interface 0 shows WAN port status and Interface 1 shows LAN port status Network Status This page shows current status of network interfaces ofthe system Type DHCP Client Mask 255 255 255 0 DNS Server 1 168 95 192 1 Interface 1 Type DHCP Server Mask 255 255 255 0 DNS Server 1 168 95 192 1 28 Bridge The Bridge setting is used to configure the Ethernet port connects to the ADSL Modem Router or Ethernet switch Bridge Settings You could configure your bridge settings in this page TCP IP Configuration IP Type Fixed IP DHCP Client PPPoE mn Mask Aa 55 255 255 0 DNS Server 1 pea PPPoE Configuration User Name Password N EE Bridge On C Off 29 Bridge When setting
7. to Bridge Mode the WAN and the LAN ports will be bridged IP Type There are three selections for Bridge Fixed IP DHCP Client and PPPoE modes For Fix IP Mode please make sure the IP address Net Mask Gateway and DNS settings are suitable in your current network environment For PPPoE Mode you have to enter correct username and password to get the IP address from your Internet Service Provider NAT Settings This embedded NAT is useful for ADSL users without NAT router and it separates the WAN port from the LAN port to perform router IP address translation Connect your PC to the LAN port set your PC as DHCP Client mode and then the PC will get an IP address from the IP Phone automatically 30 NAT Settings You could configure your NAT settings in this page u 92 168 50 136 poo Start IP T Eae Lease Time 1 0 dd hh WAN Setting IP Type Fixed IP DHCP Client PPPoE A EE Mask 0 0 0 0 DNS Server1 0 0 0 0 31 DDNS Setting DDNS Dynamic DNS A service that lets anyone on the Internet gain access to resources on your local network when the Internet address of that network is constantly changing When it detects that the IP address of the cable or DSL modem has changed it notifies the DDNS service provider of the new address DDNS Settings You could set the configuration of DDNS in this page DDNS OOn Of Host Name User Name
8. to a registered or URL Number 2 Busy Forward Activation To Enabled Disabled this function Number Forward to a registered or URL Number 3 No Answer Forward Activation To Enabled Disabled this function Number Forward to a registered or URL Number 55 4 Ring Timeout Set the Ring times to start the Forward function 2 8 Rings 2 Do not Disturb 1 Allways Block all calls 2 By Period Block calls by the period time 3 Period Time Set the start time and end time to Block calls 3 Alarm Setting 1 Activation Enable Disable alarm 2 Alarm Time Set the alarm time 4 Date Time setting 1 Date amp Time Set the IP Phone Date and Time 2 SNTP setting SNTP Enabled Disable SNTP Primary SNTP Set Primary SNTP server IP address or URL Secondary SNTP Set Secondary SNTP server IP address or URL Time zone Set Time zone Adjustment Time Set adjustment time period 5 Volume and Gain 1 Handset volume Set Handset volume from 0 15 max for you to hear 2 Speaker volume Set Speaker phone volume from 0 15 max for you to hear 3 Handset Gain Set Handset Gain from 0 15 max for remote site to hear 4 Speaker Gain Set Speakerphone Gain from 0 15 max for remote site to hear 6 Ringer 1 Ringer volume Ringer volume selection from 0 15 max 56 2 Ringer type Ringer tone selection from 1 4 7 Auto Dial Auto Dial time selection from 3 9 seconds 4 Network 1 General 1 IP Type Fixed
9. 6 ADPCM uses 4 3 or 2 bits for each sample thereby resulting in total required bandwidths of 32 000 24 000 or 16 000 bps 41 G 729 The G 729 and G 729A conjugate structure algebraic code excited linear prediction CS ACELP coding scheme also compresses PCM using advanced codebook technology It uses 8000 bps of total bandwidth G 723 The G 723 and G 723A multipulse maximum likelihood quantization MPMLQ coding schemes use a look ahead algorithm These compression schemes result in a required bandwidth of 6300 or 5300 bps GSM GSM Global System for Mobile communications is a cellular phone system standard popular outside the USA The speech signal is divided into blocks of 20 ms These blocks are then passed to the speech codec which has a rate of 13 kbps in order to obtain blocks of 260 bits Note The network administrator should balance the need for voice quality against the cost of bandwidth in the network when choosing CODECs 42 Codec ID You can setup the Codec ID in this page You need to follow the ITSP suggestion to setup these items Codec ID Setting You could set the value of Codec ID in this page Codec Type ID Default Value G726 16 ID 23 95 255 Y 23 6726 24 ID 22 95 255 22 6726 32 ID 2 95 255 M2 6726 40 ID 21 95 255 21 RFC 2833 ID 101 95 255 Y 101 Note Two VoIP devices with different Codec ID will cause the interoperability issue If you are talking with othe
10. IP client DHCP client PPPOE client 2 Fixed IP setting Host IP Subnet mask Gateway IP 3 PPPoE setting User name Password 4 DNS Server Primary DNS Secondary DNS 5 VLAN VLAN for Phone VLAN for NAT 2 Status Show IP addresses and MAC address 57 5 SIP Settings Note To set the SIP setting from keypad you have to press Menu 7 4 Administrator System Authent input the password first or the SIP setting may not be allowed to access The default password is root 1 Service Domain 1 First realm Activation User name Display name Register name Register password Proxy server Proxy Server IP Address Domain server Domain Server IP Address Outbound proxy Outbound Proxy IP Address 2 Second realm 3 Third realm 58 2 Codec 1 Codec type G 711 uLaw G 711 aLaw G 723 G 729 G 726 16 G 726 24 G 726 32 G 726 40 G 711 uLaw G 711 aLaw G 723 1 G 729A G 726 16Kbps G 726 24Kbps G 726 32Kbps G 726 40Kbps 2 VAD Voice Activity Detection Enable Disable 3 RTP Setting 1 Outband DTMF 2 Duplicate RTP No duplicate One duplicate Two duplicate Outband DTMF Enabled Disabled No resend voice packets Resend voice packets once Resend voice packets twice 4 RPort Setting RPort Enabled Disabled 5 Hold by RFC Hold by RFC3261 Enabled Disabled 6 Status Use Up Down keys to show the SIP Proxy register status 59 6 NAT Transversal 1 STUN setting 1 STUN STUN Enabled Disab
11. S N TE TE KERE 58 6 NAT Transversal es ee se ee ee ee ee ee ee ee ee ee ee ee ee ee ee ee ee 60 7 DK De Ee ee Oe OO iaa 60 5 APPLICATION EXAMPLE esse esse sessies ds se side ees si Ede se ed ee se eed esse se ede see see 61 Sk PSTN CARLING Hse EE 2 GE Ge EG GE Ge Ge NG RE ees 62 52 SIPZTO SIP CALLE Gta 63 33 SIP TO PSITN CALLING telas 65 5 4 PSTN TO SIP CALLING ccccccecececececececececececececececececececececeeeeeeeeeeesess 66 3 9 3 WAY CONFERENCE CALLING ee ee ee ee ee ee ee ee ee ee ee ee ee ee ee ee ee ee ee ee ee ee ee ee ee 67 5 6 DIRECTIPTODIRECTIPCALLING ee ee ee ee ee ee ee ee ee ee ee ee ee ee ee ee ee ee ee ee ee 69 5 7 FREEWORLD DIALUP FWD sees see ee ee se ee ee ee ee ee ee ee ee ee ee Re ee ee ee Re ee Ge ee 70 YE EE EE EE EE EE 71 Codec Setting nn OT AA OR OE EDE GEE 72 6 SPECIFICATION esse dsssde sels sede de sede se ede eed kes se gee Gee dek Ge gees de Gede ede 73 7 TROUBLE SHOOTING sssesosesssseo eeds dese dese ese dees oe esse seed de ee ede se se sies Ge de eo 75 7 1 DONOTHEARDIALTONE sesse esse ee ee ee ee ee ee ee ee ee ee ee ee ee ee ee ee ee 75 7 2 CANNOTACCESSWEBPAGE ees ee ee ee ee ee ee ee ee ee ee ee ee ee ee ee ee ee ee ee ee ee ee ee 75 1 Introduction The VOI 7000 VOI 7100 IP Phone is an LCD VoIP Phone with SIP Protocols for Voice over IP VoIP applications This user s manual will explain the keypad instructions and web configuration
12. SIP to interconnect and to establish voice sessions between each other over an IP Network SIP Telephony has emerged as a viable alternative to legacy TDM and fixed line circuits for the establishment and transmission of voice communications 35 Service Domain You may register up to three SIP accounts in the IP Phone You can call your friends via firstly enabled SIP account and receive the phone calls from all the three SIP accounts It supports 3 services allow user register on different service providers Click Active ON to enable the Service Domain then enter the following items Service Domain Settings You could set information of service domains in this page Active Oon OF Display Name User Name Register Name Register Password Domain Server Proxy Server Outbound Proxy Status Not Registered Active Oon Sor Display Name User Name Register Name 36 Realm 1 3 Active Display Name User Name Register Name Register Password Domain Server Proxy Server Outbound Proxy Status Enable the SIP account Enter the name you want to display Enter the User Name given by your ITSP Enter the Register Name given by your ITSP Enter the Register Password given by your ITSP Enter the Domain Server given by your ITSP Enter the Proxy Server given by your ITSP Enter the Outbound Proxy of ITSP If not provided you may skip this Shows register status When it shows
13. add the postfix PIN Code is used to prevent from call piracy Incorrect PIN Code will result in call disconnect If PIN code is OFF the caller may press PSTN number directly 7 Press 7654321 to call the PSTN party number of 7654321 65 5 4 PSTN to SIP Calling Applications The applications can be for ADSL connections as in both Diagrams A and B Both parties are registered to SIP server with either fixed real IP or private IP under NAT router Configurations 1 2 Same as in Example 2 Select ON in the SIP settings STUN setting page if Outbound Proxy is NOT available Select ON for the Auto Answer and PIN Code in Call settings Set the Auto Answer Ring Counter e g 3 and the PIN code e g 1234 Make sure the REG LED is ON for a successful SIP registration Callings 5 Call from PSTN line to the IP PHONE FXO number e g 7654321 Ina moment you should hear a ring back tone and wait for the IP PHONE to answer After 3 rings the VoIP mode will auto answer with a dodo tone not dial tone Press 1234 for PIN code and then you will hear a dial tone for VoIP mode Incorrect PIN Code will result in call disconnect If PIN code is OFF there will be not dodo tone and the caller may press SIP number directly Press 1688 or 1688 to call the party with the registered SIP phone number 1688 Ina moment you should hear a ring back tone and wait for the VoIP call
14. ation 5 Pick up the phone and the LCD panel will show IP Dialling and you should hear a dial tone Please hang up If not please check if the RJ45 WAN port is connected 6 Press MENU 4 Network 2 Status from the keypad to check the IP address for IP Phone The MENU key is used for escape and the ENTER key for selection The default IP address is 192 168 1 100 You need this IP address for Web configurations in Chapter 7 7 Please refer to VoIP applications examples of SIP registrations and register IP Phone into your SIP server 8 The LCD panel will show Date Time and registered lt phone number gt after successful SIP registration 9 Press the Hand Free key and you should hear a dial tone Press 123456 to call the party with the number 123456 registered in the SIP server Note that will dial out the number immediately Dialling without will not dial out until the auto dial timer default 5 seconds elapsed Ina moment you should hear a ring back tone and wait for answer 2 5 Default Setting IP Address 192 168 1 100 LAN Login Name root Password root Note WAN port IP address will depend on the device connect with For example Broadband Router provides DHCP server and assign IP address to the IP Phone when it connected 2 6 Reset to Default Press MENU 7 Administrator 2 Default setting 1 Load default by using Menu and arrow keys to reset back to factory defaults and the LCD panel will s
15. ault and the function of extension call from SIP to PSTN is disabled The FXO port is for PSTN only and no configuration is needed Calling Answering 1 Pick up the phone and press PSTN function key and you should hear a dial tone 2 Press e g 7654321 to call the PSTN party with 7654321 In a moment you should hear a ring back tone and wait for the called PSTN party to answer 3 For receiving PSTN incoming calls you just pick up the phone to answer when ringing 62 5 2 SIP to SIP Calling Applications The SIP to SIP calling works when both calling and answering parties are registered to SIP server with given registered phone numbers The ADSL connections can be as in either Diagrams Aor B Both parties are registered to SIP server under NAT router For Diagram A without NAT router you may select NAT mode to enable the embedded NAT router For Diagram B with external NAT router you may select Bridge mode to disable the embedded NAT Configurations 1 Select either NAT or Bridge in accord with your network in WAN settings page Select DHCP Client to automatically get an IP address from NAT router Remember to click the Submit button Select Active ON in the SIP settings Service Domain page gi BO ND Enter the Register Name Register Password Proxy Server and Outbound Proxy 6 Select ON in the STUN setting if Outbound Proxy is NOT available 7 Upo
16. ed party to answer 66 5 5 3 Way Conference Calling Applications The Call Transfer and 3 Way Conference Call applications are for calls among Parties A B and C Three parties are registered to SIP server with either fixed real IP or private IP There are two kinds of call transfer Blind Transfer and Attendant Transfer Blind Transfer 1 Party Acalls Party B 2 While in conversation Party B may press Transfer key and should hear a dial tone 3 Party B press Party C number and hang up to transfer to Party C Attendant Transfer 1 Party Acalls Party B 2 While in conversation Party B may press Transfer key and should hear a dial tone Party B press Party C number and talk to Party C 4 Hang up from Party B and then Party A will transfer and connect to Party C 67 3 Way Conference Call 1 Party Acalls Party B 2 While in conversation Party B may press Hold key to hold the call and should hear a dial tone Party B calls Party C 4 While in conversation Party may press Conf key to join in Party A for three way conference Call Waiting Application When a new call is coming while you are talking you will hear an interrupt dodo tone and you can press Hold key to answer the new incoming call You may press Hold key to switch back to the previous call Call Hold Application You may press Hold key to hold the current call for a while then press Hold key again to resume conversat
17. ete Selected button If you want to delete all phone numbers please click Delete All button Example Press 2 on telephone to Speed Dial the phone number 2 immediately 15 3 2 Phone Setting The sub pages are as follows Call Forward SNTP Volume Melody Ringer DND Auto Answer Dial Plan Flash Time Call Waiting Soft key Hotline and Alarm functions Phone Setting Call Forward SNTP Volume Melody Block Settings Auto Dial Call Waiting 16 Call Forward You can have your incoming calls forwarded to a specified destination You can select the forward mode and enter the forward URL Forward Setting You could set the forward number of your phone in this page All Forward of Oon Busy Forward Oof OOn No Answer Forward Oof Oon All Fwd No Busy Fwd No No Answer Fwd No No Answer Fwd Time Out 3 2 8 Ring All Forward All incoming calls are forwarded to the URL you choose Busy Forward The incoming calls are forwarded to the URL when your line is busy No Answer All incoming calls are forwarded when you Forward do not answer the call within specified time period 17 All Fwd No Specify All Forward number Busy Fwd No Specify Busy Forward number No Answer Fwd Specify No Answer Forward number No No Answer Fwd Specify the time period before forward Time Out calls Note You have to set the Time Out Timer to start to
18. ions 68 5 6 Direct IP to Direct IP Calling Applications The applications are for ADSL connection without NAT router as in Diagram A Both parties are with fixed real IP The Direct IP calling works when both calling and answering parties are with known fixed IP SIP server registrations are not required in this application Configurations 1 Select Fixed IP in the Network WAN settings page 2 Enter the items of IP Subnet Mask Gateway IP 3 Click the Submit button Callings 4 Pick up the phone and you should hear a dial tone 5 Press 211 21 191 4 or 211 21 191 4 to call the party with the real IP address of 211 21 191 4 Note that key will dial out the number immediately Dialling without will not dial out until the auto dial timer default 5 seconds elapsed In a moment you should hear a ring back tone and wait for the VoIP called party to answer 69 5 7 FreeWorld Dialup FWD Applications This shows how to use FWD as an example for free ITSP provider The applications are for both parties registered to FWD SIP server Visit FWD web site and sign up for a new registered account number Follow the instructions for registration After finished you will receive a mail sent by the FWD mail system and you will get one FWD phone number and password in the mail For example the register name phone number is 636346 with password xxxx Login to the Web configuration page FWD Web Si
19. k thee Beas tote gee ee ee 12 Speed Dial Seting RR EE KOR OR OE EE 14 3 2 PHONESETTING is sk ORE see Ee Bee EE Geb E E iin a il nn 16 NA RE EE OE OE N EE EE N 17 EE ON EO AE OER RE OE EE oe 19 Volume iii EE EE EE OE EE ON IE 20 EE N EE AE OE RE EA OE 21 BLOCKS CINE ii ET AE ET EE OS 22 Auto Answer EIE EEEE E SR ee ee RR ee ee EEEE GR ER ee ee ee Re Re ee ee 23 UP EE ER OR N ON ER ON ER EE ER EE 25 Cal Waiting EE OE RA RE RE 26 3 3 NETWORK area pie oe RE ee ee ee ds 27 Network Status RE N EE EE EO EE 28 Bridge RE eA ER HOE 29 NAT Seting Se EN EE RE EE Sone Wd EE Sata 30 AE GE OE AE ER RE EE 32 SA SIPSETIINGS a EE EE Be ee De bee 34 Service Domain sia ON heine OE EE OER ed 36 BERE Ed EERS LL OE 40 Codec RE RE EE RR ORE OE RE 43 Other Setting EO KO N OE OR OER E 44 DOs AUTOICONEIG A ee oe series 45 Auto ER se EE EE pede tae dee 45 IGMP Setting RR NEE ER EE EG 47 3 6 USER PASSWORD ioc un a Essen 47 Suk BAVEICH NGE RR EE RE EE OE EE REEN 48 3 8 UPDATE oie 48 Update DEE AE RE OE EE EE N 49 Default Setting ER AE EE GEE OO EE EG 51 3 9 REBOOT RE ER EE EE AAA AAA 51 LCD DISPLAY AND KEYPAD ussessussossossnssonsnnsnnsnnennssnssnnsonsnnssnsnnsnnnsnnne 52 4 1 KEYPAD DESCRIPTIONS iier ero i Ae near e i e ra ees 53 ADS LCD MEN Usherren e e EE EE 39 1 Phone ME Sr NE RE RE KOEN 55 2 Bes ERA GEE ORR EED RE N 55 3 Call setin ER OR EE EE EE 55 4 DNEWOTK ia nern RE RE EE ER 57 J SIP CHING OE
20. k up the phone and the LCD will show FWD phone number lt 636346 gt Press 12345 to call the party with registered FWD phone number 12345 In a moment you should hear the ring back tone and wait for the called party to answer 6 Specification Model No VOI 7000 VOI 7100 1x WAN 1 x WAN 1x LAN 1x LAN 1 x Headset Plug 1 x Headset Plug 1 x RJ11 FXO Connector SIP v1 RFC2543 v2 RFC3261 IP TCP UDP RTP RTCP IPACMP ARP RARP SNTP TFTP Client DHCP Client PPPoE Client Telnet HTTP Server DNS Client VLAN Setting DMZ Setting Virtual Server MAC Clone Setting Network Protocol Call Function Call Forward Caller ID 3 way conference 73 G 711 64k bit s PCM G 723 1 6 3k 5 3k bit s Codec G 726 16k 24k 32k 40k bit s ADPCM G 729A 8k bit s CS ACELP G 729B adds VAD amp CNG to G729 VAD Voice activity detection CNG Comfortable noise generator Voice LEC Line echo canceller Packet Loss Compensation Adaptive Jitter Buffer In Band DTMF DTMF Function Out of Band DTMF SIP Info NAT Traversal STUN Configuration Operational Temperature O to 40 C 74 7 Trouble Shooting 7 1 Do not hear dial tone When you pick up the phone and hear a busy tone it indicates the WAN port is NOT connected The LCD will show Ethernet Error Make sure the ADSL Ethernet cable is connected to the WAN port of IP Phone and Power Reset again 7 2 Can not access web page IE Web Browser is a useful
21. led 2 STUN server Server IP Address 7 Administrator 1 Auto Config 1 Config Mode Select Disable TFTP FTP HTTP for auto config function with server 2 TFTP server Set the TFTP server IP address 3 FTP server Set the FTP server IP address 4 FTP Login Name Set the login name to the FTP server 5 FTP Password Set the Password to the FTP server 2 Upgrade System You can restore to the default setting 1 Upgrade Now Select Yes No to upgrade with the upgrade Server 2 Upgrade via Select Disable TFTP FTP HTTP to do upgrade 3 Status 4 Reset Time Set Yes No to reset time 3 Default setting To load abort the default setting 4 System Authority Must enter the password first for SIP setting Default is root 5 Version This shows the firmware version 6 Watch Dog This enables Watch Dog function for debugging 7 Restart This function will restart your IP Phone 60 5 Application Example You can use PC Web browser to configure IP Phone For example enter http 192 168 1 100 from PC web browser A ADSL Connections with NAT enabled in IP Phone ADSL Modem B ADSL Connections with external NAT Router ma ADSL Modem H NAT Router 61 5 1 PSTN Calling Applications VOI 7100 is default at the VoIP mode For PSTN calls you may just pick up the phone press 0 key or PSTN function key and dial directly to the PSTN number like a normal telephone Configurations The Auto Answer is OFF at def
22. level EER E O HE E BE A one LevelOne VOI 7000 VoIP Phone VOI 7100 PoE VolP Phone User Manual Ver 2 4 1008 Safety FCC WARNING This equipment may generate or use radio frequency energy Changes or modifications to this equipment may cause harmful interference unless the modifications are expressly approved in the instruction manual The user could lose the authority to operate this equipment if an unauthorized change or modification is made This equipment has been tested and found to comply with the limits for a Class B digital device pursuant to Part 15 ofthe FCC Rules These limits are designed to provide reasonable protection against harmful interference in a residential installation This equipment generates uses and can radiate radio frequency energy and if not installed and used in accordance with the instructions may cause harmful interference to radio communications However there is no guarantee that interference will not occur in a particular installation If this equipment does cause harmful interference to radio or television reception which can be determined by turning the equipment off and on the user is encouraged to try to correct the interference by one or more of the following measures 1 Reorient or relocate the receiving antenna Increase the separation between the equipment and receiver 3 Connect the equipment into an outlet on a circuit different from that to which the recei
23. ly convert analog signals to digital form and vice versa Codec Settings You could set the codec settings in this page Codec Priority 1 6 711 u law 7 Codec Priority 3 G 723 E Codec Priority 5 G 726 16 vw Codec Priority 7 G 726 32 Codec Priority 9 GSM v 6 711 amp 6 729 Er A G 723 5 3K 6 723 5 3K Oon of Voice VAD Voice VAD OOn of 40 Codec Priority Adjust Codec priority to meet your requirement lower number shows higher priority RTP Packet Adjust Codec g711 9729 and g723 packet Length length G 723 5 3K Enables 5 3K bit s rate when use 9723 Voice VAD VAD Voice Activity Detection is used to reduce the transmission rate during inactive speech periods VAD classifies the input signal into active speech inactive speech or background noise Based on the VAD decisions One of the most important factors is how much bandwidth is used for each VolP call The higher the CODEC bandwidth is the higher the cost of each call across the network will be Following is a list of CODECS and their associated bandwidth G 711 The G 711 pulse code modulation PCM coding scheme uses the most bandwidth G 711 takes samples 8000 times per second each of which is 8 bits in length for a total bandwidth of 64 000 bps G 726 The G 726 adaptive differential pulse code modulation ADPCM coding schemes use somewhat less bandwidth While each coding scheme takes samples 8000 times per second like PCM G 72
24. n successful SIP registration the REG LED indicator will be ON and the LCD will show registered lt phone number Callings 8 Pick up the phone and you should hear a dial tone for VoIP mode 9 Press 1688 or 1688 to call the party with the registered SIP phone number 1688 Note that key will dial out the number immediately 63 Dialling without will not dial out until the auto dial timer default 5 seconds elapsed 64 5 3 SIP to PSTN Calling Applications The SIP to PSTN calling works when both calling and answering parties are registered to SIP server with given registered phone numbers The ADSL can be as in both Diagrams Aand B Both parties are registered to SIP server with either fixed real IP or private IP under NAT router Configurations 1 Same as in Example 2 2 Select ON in the SIP settings STUN setting page if Outbound Proxy is NOT available 3 Select ON for the Auto Answer and PIN Code in Call settings Set the Auto Answer Ring Counter e g 3 and the PIN code e g 1234 4 Upon successful SIP registration the REG LED indicator will be ON Callings 5 Pick up the phone for VoIP mode and press 1688 or 1688 to call another IP Phone with registered SIP phone number 1688 6 After 3 rings for Auto Answer the FXO port will auto answer with a dodo tone not dial tone Press 1234 for PIN code and then you will hear a PSTN dial tone Note you must
25. or the incoming call from PSTN the FXO port will answer with a short beep tone and allow caller to redial to VoIP number PIN Code is used to prevent from call piracy The caller needs to enter the right PIN code followed by to get the PSTN dial tone Incorrect PIN Code will result in call disconnect The Auto Answer is disabled at default Auto Answer You could enable disable the auto answer in this page Auto Answer OOn Of Auto Answer Counter 0 8 PIN Code Enabled OOn OF PIN Code 23 Auto Answer Auto Answer Counter PIN Code Enabled PIN Code Enable this function to answer the incoming calls from PSTN line automatically It allows user to place call to Internet again Set time period before phone pick up the calls automatically Enable the call restriction from PSTN line to VoIP or vice versa Set the PIN code User requires to enter correct code which correspond with before get second dial tone Note This function is only available on VOI 7100 24 Auto Dial Auto dial timer settings can be set in this page The auto dial timer specifies the elapse time between the dialling digits Auto Dial Setting You could the time slice to auto dial in this page Auto Dial Time 5 3 9 sec Submit Reset Auto dial Time The inter digit timer Default is 5 seconds 25 Call Waiting You can enable the call waiting function in this page It allows answering another coming call by
26. reboot with the stored configurations Reboot System You could press the reboot button to restart the system Reboot system 51 4 LCD Display and Keypad You can use keypad to configure and to check the status of IP Phone Make sure that the WAN port is connected to ADSL Ethernet or you may hear a busy tone from the telephone 52 4 1 Keypad Descriptions a b c A B C az p q g P Q R S yP y z Start dialling process 53 The Enter is for setting selections The Menu key is to set the IP Phone ae Use navigate keys to select menu items VOL Set the volume High Low TRANSFER Transfer to the other phone number DEL To delete parameters while change the setting This is for Speaker Phone These are for 6 speed dial numbers 54 4 2 LCD Menu 1 Phone Book 1 Search Search Phone Book 2 Add entry Add new phone number to phone book 3 Speed dial Add speed dial phone number 4 Erase all Erase all phone number 2 Call History 1 Incoming calls Show all incoming call 2 Dialed numbers Show all dialled call 3 Erase record Delete call history 1 All Delete all call history 2 Incoming Delete all incoming call 3 Dialled Delete all dialled out call 3 Call setting 1 Call forward 1 All Forward Activation To Enabled Disabled this function Number Forward
27. rs got some problems you may ask the other one what kind of 43 Codec ID he use then you can change your Codec ID Other Settings You can setup the Hold by RFC and QoS in this page To change these settings please follows your ITSP information The OoS is used to set the voice packet priority Higher value other than zero will get higher priority for the voice packets in Internet However the QoS function still needs to cooperate with the other Internet devices SIP Expire Time depends on your ITSP required Other Settings You could set other settings in this page Hold by RFC OOn Gor Voice QoS Dif Sen 40 0 63 SIP QoS DiffServ 40 j 0 63 SIP Expire Time 60 15 86400 sec Use DNS SRY OOn of Note For more information about these advanced features please ask your network administrator or service provider help desk 44 3 5 Auto Config Auto Configuration function can be used to download the original configurations stored in the TFTP or FTP server Auto Config Auto Config ICMP Settings Auto Config This feature allows service provider to provision their customer s IP Phone end to end By employing a TFTP FTP server the provisioning server writes the configuration files needed to automatically configure the IP Phone Before enabling this auto configuration you must select Bridge ON and Fixed IP type in Network settings 45 Auto Configuration Setting
28. s for the VoIP Phone IP Phone can make a VolP call over the ADSL Internet connection and it provides one RJ45 WAN port for ADSL Internet connections plus one RJ45 LAN port for Notebook PC connection With the embedded NAT DHCP server IP Phone can be easily configured for different network diagrams by PC Web browser and telephone keypads This is very suitable for ITSP Internet Telephony Service Providers and SOHO users to make VolP calls Moreover with PPTP VPN client supported user can create secured tunnel between central office and IP Phone make sure your communication is safe VOI 7100 IP phone adopts the latest Power Over Ethernet technology to not only save user investments simplify network deployment but also provide centralized power management 1 1 Features 7 SIP v1 RFC2543 v2 RFC3261 with MD5 authentication RFC2069 and RFC 2617 z RJ45 x 2 for Ethernet WAN and LAN ports ITU T G 711 G 723 G 726 G 729A B VAD and CNG for Speech Codec ITU T G 165 168 Echo Cancellation 7 LCD Display for registered IP phone number Configurations by Web Browser and Telephone Keypads 7 Embedded NAT DHCP Server 7 PPPoE DHCP Client for Dynamic IP plus NAT DNS and DDNS Clients Support STUN server for NAT Traversal Speed Dial Call Forward Waiting Transfer Hold and 3 Way Conference Call features 7 Direct IP URL Dial without SIP Proxy or Dial number via SIP server 7 Phone book stores up to 140 records VPN PPTP
29. t the volume of ringer Set the volume send out to the other side s handset Set the volume send out to the other side s speaker 20 Melody You may set ON the ringer and select different ringer type for Melody settings R nger Settings You could set your favorite ringer in this page Ringer Oon Gor Ringer Type ringer 1 Y Note Because the default ringer is ringer 1 it means the setting will remain as off if you switch On and select ringer 1 21 Block Setting You can setup the Block Setting to keep the phone silence You may set this feature when you are in a meeting or busy Block Setting You could set the block period of your phone in this page Always Block C On Off Block Period On Off From Joo foo hh mm To oo 00 hh mm Submit Reset Always Block All incoming call will be blocked when enabled Block Period Set a time period and the phone will be blocked during the time period When the time in From is greater than To the Block time will be from Day 1 to Day 2 22 Auto Answer You may enable the Auto Answer function to answer the incoming call by FXO port When the ring count exceeds the number set in Auto Answer Counter the FXO port will auto answer and allow for extension calls from PSTN to VoIP and vice versa For the incoming call from the Internet the FXO port will answer with a PSTN dial tone and allow caller to redial to PSTN phone number F
30. tart showing Loading Program and System Initialized Please use the MENU key for escape and the ENTER key for selection Press MENU 7 Administrator 6 Restart to reboot IP Phone 3 Web Configuration You may enter the IP address from PC Web browser to configure IP Phone For example enter http 192 168 1 100 from Web browser to display login page as follows Login VoIP Gateway Username root Password e C Remember last login Enter the username and password into the blank field The default settings are Username root Password root Click the Login button will enter the management information page for system setup Note Whenever you change the setting in each Web page please remember to click the Submit button in the page and click the Save button to save into the non volatile memory and click the Reboot button to activate the new settings 10 System Information After login you will see the system information like firmware version Codec etc in this page You may click the button list at the left hand side to configure the IP Phone o lt o ONEEE JEE E DEEEE Phone Book Phone Setting Network SIP Settings Others User Password Save Change Update Reboot System Information This page illustrate the system related information Model Name VOIP PHONE Firmware Version Tue Jun 12 17 42 50 2007 O1ug_o Codec Version Thu Apr 19 13 59 33 2007 Cau
31. te http www freeworlddialup com Username FWD Number Password FWD Password Domain fwd pulver com SIP Proxy fwd pulver com 5060 Outbound Proxy fwdnat pulver com 5082 OR STUN server stun fwdnet net 3478 Phone must be STUN enabled Listen RTP Port 8000 Listen SIP Port 5060 70 SIP Settings Realm eC Active On OOR Display Name 636346 User Name 636346 Register Name 636346 Register Password s Domain Server fwd pulver com Proxy Server fwd pulver com 5060 Outbound Proxy fwdnat pulver com 5082 Status Not R gisiered You have to enter the Display Name User Name Registered Name Registered Password Domain Server Proxy Server Outbound Proxy After finished the setting click the Submit button and the Save Change button The IP Phone will reboot automatically After boot up the SIP setting page will show Registered and the LCD will show registered lt phone numbers it will shows No service otherwise 71 Codec Setting Codec Settings You could set the codec settings in this page Codec Priority Codec Priority 1 G 729 v Codec Priority 2 G 723 v Codec Priority 3 S711 wlaw y Codec Priority 4 6 211 alaw x Codec Priority 5 G 726 16 v Codec Priority 6 G 726 24 y Codec Priority 7 G725 32 w Codec Priority 8 G 726 40 v 6 711 amp 6 729 20ms 6 723 30ms vi G 723 5 3K G 723 5 3K OOn Gor Voice VAD Voice VAD O0n Gor Callings 1 2 Pic
32. tion VOI 7000 and VOI 7100 use different firmware format check it carefully before upgrade 3 1 Phone Book The Phone Bock specifies pre record phone list and speed dialling function it allows up to 140 records on the phone book Phone Book You could add delete items in current phone book Phone Book Page Phone U Name U URL Select E O A SS EA A 2 i 4 E 6 DO E am 8 O o EI Add New Phone Position 0 139 URL 12 Input the Position 0 139 Name and URL then click the Add Phone button to enter Note URL can be either complete strings or numbers only it depends on your service provider Example Li ET EIA ES 1 David 221 a 2 Bill 221090 sipcall org a 3 Jone 221080 192 168 12 234 o 4 13 Speed Dial Setting For Speed Dial function you can add delete Speed Dial number up to maximum 10 entries in Speed Dial Phone List Speed Dial Phone List You could set the speed dial phones in this page Phone Name CC Select 0 O E O 5 4 en u A lS EA A J WEE O Ee Ee E Add New Phone Position 0 9 14 If you need to add a phone number into the Speed Dial list you need to enter the position the name and the phone number by URL type When you finished a new phone list just click the Add Phone button If you want to delete a phone number please select the phone number you want to delete then click Del
33. tool to configure IP PHONE When you have difficulties in accessing the default IP address http 192 168 1 100 of IP PHONE as in the following figure the most possibility is that the PC might have different subnet IP settings from 192 168 1 xxx In this case you must change IP PHONE IP address to the same subnet as PC and NAT router ma ADSL Modem H NAT Router JD Phone IP 192 168 1 100 75 Example To change IP PHONE IP address to the same subnet as PC and NAT router 1 Press the menu to enable DHCP Client mode IP PHONE will reboot and LED will start flashing to get an IP address from NAT DHCP server Press Menu 4 5 to read IP Addresses for WAN and LAN Ports for example 192 168 62 51 Enter from IE web browser http 192 168 62 51 to login IP PHONE web page for configurations 76
34. upgrade Update New Firmware Default Caution VOI 7000 and VOI 7100 use different firmware format check it carefully before upgrade 48 Update Firmware Update Firmware You could update the newest firmware Method LocalPC OTFTP Local PC Code Type Risc File Location TFTP TFTP Server 192 168 1 250 The IP Phone provides two methods HTTP or TFTP to update new firmware as the following steps 1 2 Select the firmware code type Risc or DSP code mostly for Risc code Click the Browse button to choose the updated file location for HTTP download or Select TFTP and enter the IP address of TFTP server for firmware download then click the Update button Caution VOI 7000 and VOI 7100 use different firmware format check it carefully before upgrade Do Not power off during the upgrade processing it may damage the IP Phone For update firmware by TFTP the TFTP server is required Contact your network administrator for more information 50 Default Setting You can restore the IP Phone to factory default in this page By clicking the Restore button the IP Phone will restore to default and automatically restart again Restore Default Settings You could click the restore button to restore the factory settings Restore default settings 3 9 Reboot You may click the Reboot button to restart then IP Phone will automatically
35. ver is connected 4 Consult the dealer or an experienced radio TV technician for help CE Declaration of conformity This equipment complies with the requirements relating to electromagnetic compatibility EN 55022 class B for ITE the essential protection requirement of Council Directive 89 336 EEC on the approximation of the laws of the Member States relating to electromagnetic compatibility Efe E 1 Table of Contents INTRODUCTION issie osse ees ek seen god dose N Gog ee ek don dose es do ego AR 1 MED FEATURES EE ME ET ER enter 2 1 2 BACKINGCONTENTS n ne nur Ge Ge GEVERG ee gee eg 3 HARDWARE DESCRIPTION sesse sesse see se ese see se ese se Se ee se Se Ee Ge Se Ee Ge Se ee ee 4 2 1 LCD DISPLAY AND KEYPADS ie esse ee se se es se ee ee se ee ee ee ge ee ee Ge Re ee ee GR Re ee ee 4 2 2 ERONTPANED SEGE ee ee de Ese Rd SWEER srl Ee GE See be be ig 5 2 3 CONNECTION DIAGRAM soseen see ese ee ae ese ae ee Ge Ge Ge ee Se Ge ee Se 7 Without Broadband Router ies se ee ee se Ee ee ee Re Re ee Re Re ee ee ee ee ee 7 With Broadband Router ees se se ee ER Re Re RR Re Re ee ee ee Re Re ee ee ee ee ee 7 24 INSTALLATION nennen Ge See ee See Een ee eed 8 2 2 DEFAULT SETTING acia twa GE Gee ee ee De 9 2 6 RESET TO DEFAULT 95526 sic iese Ee Re EER Ska gee ees ee GR EER RE GER ee See 9 WEB CONFIGURATION esse sesse see sesse see se sae Be Ge GE Be Ee GE Be Ge Ge Be Ee Ge Se ee 10 Bids PHONE BOOK 2 u Rent cas
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