LevelOne VOI-7010 LCD Wired handset Grey
Contents
1. 4440022 25 Auto Answer eese eene nennen nennen nnne 26 Dial Plat i Ee tae 28 utet uote is 33 ioni Ness 34 SOfUtKOy scare estt Psi 35 Hot litigio et eet det teer eeu es 36 Alarm itte e e ete te eo eii 37 9 39 NETWORK suba ana hed 38 Network 39 WAN Settings e Reed etudes 40 LAN Settings i tette 42 DDNS Setting 43 5 aia 45 47 Virtual SERVER 48 meten tcu um a deena 50 3 4 52 Service Domain ien eei 54 COJO ae 58 Code Di iine et eee Gott 61 Other Settings piede ten eds 62 3 5 OTHERS oe tg eae de e Vendee a Da adu e 63 Auto Config oit ete tints 64 saith eet eee en ee is 65 MAG Clone ebrei 66 2 UTILES e ht took ote tel 67 a hee aise E 68 3 6 SEREPASSWORD3 I pra pork de 70 3 7 70 3 8 UPDATES eee eit thei eite 71 Update nn 72 Auto Update 2 74 Default 75 3 95 R2. MENU ENTER REG VMS A f 1 Vokt UP DOWN m P BOOK 106 DEL MUTE CONF WEIL gt ERI Y j 2 dus TON M7 5 6 PSTN Omo S n es _ VMs x ES E N 2 SS 2 22 77 4 1 Keypad Descriptions Descriptions 0 Space 78 79 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 80 3 Call setting 1 Call forward 1 All Forward Activation Enabled Disabled this function Number Forward to a registered or URL Number 2 Busy Forward Activation Enabled Disabled this function Number Forward to a registered or URL Number 3 No Answer Forward Activation Enabled Disabled this function Number Forward to a registered or URL Number 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
3. E mail Address DDNS Server DDNS Server List Type Wild Card Off Off Line On Off 43 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 Host Name User Name Password E mail Address DDNS Server DDNS Server List Type Wild Card BACKMX Off Line GOn Oof levelone dyndns org levelone members dyndns org dyndns on v of In this example the other user can place VoIP calls to your IP Phone directly by your domain address 44 VLAN Settings This function provides packets control over LAN it must work with Ethernet switch supported 802 1Q compliant can be configured to transmit tagged or untagged frames A tag field containing VLAN and or 802 1p priority information can be inserted into an Ethernet frame VLAN Settings You could set the VLAN settings in this page VLAN Packets OO Gof VID 802 1 Q TAG 136 2 4094 User Priority 802 1P 0 0 27 CFI 1 0 1 VLAN Packets If you enable VLAN Packets and set the VID User Priority and CFI then all the incoming packets will be checked with the IP Address and the VID VID VLAN identifier VID between one and 802 1Q TAG 4 094 into each f
4. 0 10 Speaker Gain 5 0 10 PSTN In Gain 5 0 10 22 Handset Vol Speaker Vol Ringer Vol PSTN Out Vol Handset Gain Speaker Gain PSTN In Gain Set the volume to hear from the handset Set the volume to hear from the Speaker Set the volume of ringer Set the PSTN volume for you to hear Set the volume send out to the other side s handset Set the volume send out to the other side s speaker Set the volume send out to the other side s handset Note PSTN function is only available on VOI 7011 23 Ringer You may set ON the ringer and select different ringer type for Melody settings Ringer Settings You could set your favorite ringer in this page Ringer OOn Gof Ringer Type 1 Note Because the default ringer is ringer 1 it means the setting will remain as off if you switch On and select ringer 1 24 DND Do Not Disturb You can setup the DND Do Not Disturb to keep the phone silence You may set this feature when you are in a meeting or busy DND Setting You could set the do not disturb period of your phone in this page DND Always DND Period From To DND Always DND Period OOn Gof hh mm hh mm All incoming call will be blocked when enabled 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
5. 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 X Table of Contents 1 3 INTRODUDGCTIQON eric nene du oe 1 2 1 2 PACKING 3 123 OPTIONAL A 3 HARDWARE DESCRIPTION 5 2 1 LCD DISPLAY AND 9 5 2 2 FRONT PANEL tren Rer de ER eee est 6 2 3 CONNECTION DIAGRAM eene 8 2 4 INSTALLATION uice ete inte nee 9 2 5 DEFAULT SETTING essen 10 2 6 RESET TO DEFAULT 10 WEB CONFIGURATION ee ererer nnne nennt 12 3 1 PHONE BOOK ius reete 14 Speed Dial Setting sse 16 3 2 PHONE SETTING treten detenta 18 Call 0 0000 0000100 19 21 Melun 22 RING pe 24 Do Not
6. 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 86 5 Application Example You can use PC Web browser to configure IP Phone For example enter http 192 168 123 1 from PC web browser A ADSL Connections with NAT enabled in IP Phone ADSL Modem ADSL Connections with external NAT Router a ADSL Modem E NAT Routere 87 5 1 PSTN Calling Applications VOI 7011 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 default 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 ca
7. elapsed 90 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 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 91 you must 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 Press 7654321 to call the PSTN party number of 7654321 92 5 4 PSTN to SIP Calling Applications The applications can be for ADSL connections as in both Diagrams Both parties are registered to SIP s
8. 0 lt uM E ommum LevelOne VOI 7010 VOI 7011 SIP IP Telephone User Manual Ver 1 1 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 of the 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 2 Increase the separation between the equipment and receiver 3 Connect the equipment into an outlet on a circuit different from that to which the receiver is connected
9. 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 forward the calls It requires Submit Save and Reboot to activate new settings 20 SNTP You can setup the primary and second SNTP Server IP Address to get the date time information You may also set the 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 Oof Primary Server time window com Secondary Server 208 184 49 9 Time Zone GMT 08 vi 00 v hh mm Sync Time 1 50 20 84 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 21 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 5 0 10 Speaker Volume 5 10 10 Ringer Volume g 0 10 PSTN Out Volume 5 0 10 Handset Gain
10. Initiation Protocol 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 53 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 Realm OCS Active Gof Display User Register Register Password Domain Server Proxy Server Outbound Proxy Status Not Registered Active Gof Display Name User Name Register Name 54 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 sk
11. 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 38 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 of the system Interface 0 Type DHCP Client Mask 255 255 255 0 DNS Server 1 158 95 192 1 Type DHCP Server Mask 255 255 255 0 DNS Server 1 168 95 192 1 39 WAN Settings The WAN setting is used to configure the Ethernet port connects to the ADSL Modem Router or Ethernet switch WAN Settings You could configure the WAN settings in this page LAN Mode OBridge O NAT IP Type OFixedIP 9 DHCP Client O PPPoE 51 Mask 255 255 255 0 DNS 168 95 192 1 MAC 101168013387 User Name m Service Name 40 LAN Model IP Type The default setting is NAT mode for IP Phone and this enables the embedded NAT router between the LAN port and PC port You may change to Bridge Mode if you need NOT use the embedded NAT router When setting to Bridge Mode the WAN and the LAN ports will be bridged There are three selections for WAN Fixed IP DHCP Client and PPPoE modes This WA
12. 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 81 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 2 Ringer type Ringer tone selection from 1 4 7 Auto Dial Auto Dial time selection from 3 9 seconds 82 4 Network 1 WAN Setup 1 IP Type Fixed IP client DHCP client PPPoE client 2 Fixed IP setting Host IP Subnet mask Gateway IP PPPoE setting User name Password 2 LAN Setup 1 Bridge 2 NAT 3 DNS Server 1 Primary DNS 2 Secondary DNS 4 VLAN 1 Activation 2 VID 3 Priority 4 CFI 5 Status Show IP addresses of WAN LAN and MAC address use UP Down keys 83 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
13. 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 84 2 Codec 1 Codec type G 711 uLaw G 711 uLaw G 711 aLaw G 711 aLaw G 723 G 723 1 G 729 G 729A G 726 16 G 726 16Kbps G 726 24 G 726 24Kbps G 726 32 G 726 32Kbps G 726 40 G 726 40Kbps 2VAD Voice Activity Detection Enable Disable 3 RTP Setting 1 Outband Outband DTMF Enabled Disabled 2 Duplicate RTP No duplicate No resend voice packets One duplicate Resend voice packets once Two duplicate 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 85 6 NAT Transversal 1 STUN setting 1 STUN STUN Enabled Disabled 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 Setthe 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
14. to Day 2 25 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 For 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 Auto Answer Counter 0 8 PIN Code Enabled Gof PIN Code 26 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 7011 27 Dial Plan Dial plan and
15. 2 17 42 50 2007 0109 o Codec Version Thu Apr 19 13 59 33 2007 Network SIP Settings Others User Password Save Change Update Reboot Caution VOI 7010 and VOI 7011 use different firmware format check it carefully before upgrade 3 1 Phone Book The Phone Book 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 ee SS Add New Phone Position 0 139 URL 14 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 7 1 David 221 2 Bill 221090 sipcall org 3 Jone 221080 192 168 12 234 4 15 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 0 O SE E 2 4 BA lll he Delete Selected Add New Phone Position 0 9 URL 71 16 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 finis
16. EBOOT dt 76 4 LCD DISPLAY AND KEYPAD 77 4 1 KEYPAD 86 2 444 4244 78 4 2 0 80 1 Phone Book nes 80 2 Gall ElStOEy i e perte itai 80 3 Call 81 4 Network 83 5 SIP Settirids x iine Ire ek ee 84 6 NAT 0402002 0 86 T Administrator ce tete 86 5 APPLICATION 87 5 1 PSTN CALLING ecce eene nennen 88 5 2 SIP TO SIP CALUNG 89 5 3 SIP TO PSTN 91 5 4 PSTN TO SIP CALLING eene 93 5 5 3 WAY CONFERENCE 95 5 6 DIRECT IP TO DIRECT IP 97 5 7 FREEWORLD DIALUP FWD 98 SIP Settings ne eec dee 99 Codec Setting rc ced enean 100 6 5 101 7 TROUBLE 5 103 7 1 DO NOT HEAR DIAL 103 7 2 CAN NOT ACCESS WEB 103 1 Introduction The VOI 7010 VOI 7011 IP Phone are an LCD Vo
17. IP Phone with SIP Protocols for Voice over IP VoIP applications IP Phone can make a VoIP 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 VoIP 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 7011 provides one FXO port to connect to traditional telephone line it allows forwarding calls from traditional telephone line to VoIP or vice versa 1 1 Features SIP v1 2543 v2 RFC3261 with MD5 authentication RFC2069 and RFC 2617 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 a LCD Display for registered IP phone number Configurations by Web Browser and Telephone Keypads Embedded NAT DHCP Server 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 Phone book s
18. N setting is for the LAN port when set in NAT mode The WAN default is at DHCP Client Mode 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 41 LAN 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 LAN Settings You could configure the LAN settings in this page LAN Setting IP 192 168 123 1 Mask 255 255 255 0 MAC DHCP Server DHCP Server 0O0n Start IP 150 End IP 200 Lease Time 1 10 dd hh Reset Note You must set LAN Mode as NAT under WAN Settings otherwise the DHCP Server will not work 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 off Host Name User Name Password
19. User 1636346 Register Name 1636346 Register Password Domain Server fwd pulver com Proxy Server fwd pulver com 5060 Outbound Proxy fwdnat pulver com 5082 Status Not Fegistered 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 phone numbers it will shows No service otherwise 99 Codec Setting Callings Pick up the phone and the LCD will show FWD phone number lt 636346 gt 1 Codec Settings You could set the codec settings in this page Codec Priority Codec Priority 1 G 729 v Codec Priority 2 G73 Codec Priority 3 G71 whaw Codec Priority 4 711 Codec Priority 5 06 16 v Codec Priority 6 07 4 vw Codec Priority 7 G 726 32 Codec Priority 8 9726 40 v RTP Packet Length 0 711 amp 0 729 Wms v G 723 30ms G 723 5 3K G 723 5 3K Gof Voice VAD Voice VAD Gof Press 12345 to call the party with registered FWD phone number 12345 n a moment you should hear the ring back tone and wait for the called party to answer 100 6 Specification Model No VOI 7010 VOI 7011 1x WAN 1 x WAN 1 x LAN 1 x LAN Connector 1 x Heads
20. ate settings in this page Update via Cor OHTIP Server Exp download FTP Username FTP File Path Exp filefload Check new firmware Scheduling Scheduling Time AM 00 00 05 59 Firmware File Prefix PHONE Next update time 74 Note This function is mainly for your ISP settings only ask your network administrator before change any parameters 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 Restore 75 3 9 Reboot You may click the Reboot button to restart then IP Phone will automatically reboot with the stored configurations Reboot System You could press the reboot button to restart the system Reboot system 76 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 0 n PICK UP _REDIAL HOLD TRANSFER
21. auto dial timer settings can be set in this page The dial plan allows you to map the dialling into an easy to remember phone number system The auto dial timer specifies the elapse time between the dialling digits Dial Plan You could the set the dial plan in this page Drop prefix Yes Replace rule 1 T Drop prefix O Yes Replace rule 2 O Yes Replace rule 3 Drop prefix OYes Replace rule 4 EN Dial now Auto Dial Time 5 8 9 Use as send key S Yes ONo Use for IP dialing Yes ONo 28 When Drop prefix is ON and the dialling prefix is matched the prefix will be dropped and replaced by the rule digits and followed by the rest of dialling digits When Drop prefix is OFF and the dialling prefix is matched the rule digits will be added before the dialling digits in accord with the settings Dialling Prefix Example Drop prefix OYes No Replace rule 1 002 58613 5662 Drop prefix Yes Replace rule 2 006 002 003 004 005 007 009 Drop prefix O Yes No Replace rule 3 009 112 Drop prefix OYes Replace rule 4 007 5 21 Dial now 11 Auto Dial Time 3 9 sec Use as send key 9 Yes Use for IP dialing Yes O No Note Symbol x equals 0 1 2 3 4 5 6 7 8 9 Symbol equals or 29 Example 1 D
22. d to local servers in the LAN It also capable of port redirection when incoming traffic to a particular port may be redirected to a different port on the server computer Virtual Server Settings You could set your virtual servers in this page The usual port numbers are WEB TCP 80 FTP Control TCP 21 FTP Data TCP 20 E mail POP3 TCP 110 E mail SMTP TCP 25 DNS UDP 53 and Telent TCP 23 Virtual Server Page page 1 Enable Protocol inPort ExPort Select Enable Selected Delete Selected Delete All Reset Add Virtual Server Num 0 23 Server IP Protocol TCP Internal Port External Port 48 For example if use runs ftp server on the LAN IP address is 192 168 1 8 port number is 21 as ftp standard In this case you can access your local network ftp server via Internet through Virtual Server enabled IP Phone Virtual Server Page page 1 Num enable Protocol In Port SeveriP Select lect 0 TCP 21 21 192 168 1 8 1 2 3 4 5 6 7 Enable Selected Delete Selected Delete All Application Diagram IP Phone ftp server 49 PPTP Point to Point Tunnelling Protocol PPTP is a network protocol that enables the secure transfer of data from a remote client to a private enterprise server by creating a virtual private network VPN across TCP IP based data networks PPTP supports on de
23. efault 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 start 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 10 3 Web Configuration You may enter the IP address from PC Web browser to configure IP Phone For example enter http 192 168 123 1 from Web browser to display login page as follows Login VoIP Gateway Username root Password eee 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 12 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 level Hae ON a En System Information This page illustrate the system related information Model Name VOIP_PHONE Phone Setting Firmware Version Tue Jun 1
24. erver with either fixed real IP or private IP under NAT router Configurations 4 Same as Example 2 2 Select ON in the SIP settings STUN setting page if Outbound Proxy is NOT available 3 Select 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 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 In a 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 93 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 In a moment you should hear a ring back tone and wait for the VoIP called party to answer 94 5 5 3 Way Conference Calling Applications The Call Transfer and 3 Way Conference Call applications are for calls among Parties A B andC 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 2 Party Acalls Par
25. et Plug 1 x Headset Plug 1xRJ11 FXO SIP v1 RFC2543 v2 RFC3261 IP TCP UDP RTP RTCP IP ICMP ARP RARP SNTP TFTP Client DHCP Client PPPoE Client Telnet HTTP Server Network Protocol DNS Client VLAN Setting DMZ Setting Virtual Server MAC Clone Setting Call Hold Call Waiting Call Function Call Forward Caller ID 3 way conference 101 G 711 64k bit s PCM G 723 1 6 3k 5 3k bit s G 726 16k 24k 32k 40k bit s ADPCM G 729A 8k bit s CS ACELP G 729B adds VAD amp CNG to G 729 VAD Voice activity detection CNG Comfortable noise generator 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 0 to 40C Temperature 102 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 tool to configure IP PHONE When you have difficulties in accessing the default IP address http 192 168 123 1 of IP PHONE as in the following figure the most possibility is that the PC might have different subnet IP settings from 192 168 123 xxx In this case you must change IP PHONE IP address to the same s
26. gory 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 TheLCD panel will show Date Time and No service without SIP registration or phone number after successful SIP registration 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 192 168 123 1 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 phone number 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 9 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 192 168 123 1 LAN Login Name root Password root 2 6 Reset to D
27. hanges You have to save changes to effect them Save Changes 70 3 8 Update User can update the IP Phone firmware when new firmware is available Make sure no power off during the firmware upgrade Update New Firmware Auto Update Default Caution VOI 7010 and VOI 7011 use different firmware format check it carefully before upgrade 71 Update Firmware Update Firmware You could update the newest firmware Method Local PC Local PC Code Type Risc File Location TFTP TFTP Server 192 168 1 250 Reset The IP Phone provides two methods HTTP or TFTP to update new firmware as the following steps 1 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 7010 and VOI 7011 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 73 Auto Update Settings The IP Phone provides three methods TFTP FTP or HTTP to update new firmware as the following steps Auto Update Settings You could set auto upd
28. he 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 726 ADPCM uses 4 3 or 2 bits for each 59 sample thereby resulting in total required bandwidths of 32 000 24 000 or 16 000 bps 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 i
29. hed 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 Delete 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 17 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 DND Auto Answer Dial Plan Flash Time Call Waiting Soft key Hot line Alarm 18 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 Busy Forward Oof Oon No Answer Forward of All Fwd No Busy Fwd Na No Answer Fwd No No Answer Fwd Time Out 2 8 Ring Reset 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 19 All Fwd No Specify
30. installed by your ISP and replace the WAN MAC address with the MAC address of the IP Phone MAC Clone Setting You could enable disable the MAC clone setting in this page MAC Clone GOn Oof Note It is not recommended that you change the default MAC address unless required by your ISP 66 Tones The Tone setting can be adjusted to generate Dial tone Ring tone Ring Back tone and Busy tone for different countries Tones Settings You could configure your tones settings in this page Dial Ring Back 2 2 NE IIIS v v v v Cadence On Hi Tone Freq 440 480 520 520 480 440 Lo Tone Freq 350 440 480 480 440 350 Hi Tone Gain 4522 2261 2261 2261 15360 2261 Lo Tone Gain 2261 2261 2261 2261 16360 1130 On Time 1 200 50 30 200 30 Off Time 1 0 400 50 20 400 20 On Time 2 0 0 0 0 0 30 Off Time 2 0 0 0 0 0 400 3 0 0 0 0 0 0 Off Time 3 0 0 0 0 0 0 Note To meet your current system tone settings please refer to PBX technical manual or ask telecom technician Advanced The advanced sett requirements The ings might be useful for some network ICMP function is to echo when someone ping this device This can prevent from hacker attacking the device by not echo ing Advanced Setting You could change advanced setting in this page ICMP Not Echo Send Anonymous CID Management from WAN Send Flash event SIP Encrypt PPPoE retry per
31. iod System Log Server System Log Type ICMP Not Echo O Yes No QYes No Yes ONo Disabled H Disabled v 5 Seconds None v 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 68 Send Anonymous CID Management from WAN Send Flash Event SIP Encrypt PPPoE Retry Period System Log Server System Log Type Select No if you subscribe to CallerID service on your PSTN line otherwise Yes Select Yes to allow user manage the IP Phone from WAN Select DTMF Event the Flash will be sent as a DTMF event Select SIP INFO Flash is transmitted by SIP INFO messages This feature only work with ITSP required Set Re connect time period when DSL PPPoE connection is disconnected Enter IP address or URL of log server To set the log type it depends on your network administrator requirement 69 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 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 Q N Save C
32. ip this Shows register status When it shows 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 55 TMF Setting 2833 O Inband Send DTMF SIP Info SIP Port 5060 1024 65535 RTF Port 60000 1024 65535 STUN Setting STUN Gof STUN Server stun xten com STUN Port 3478 MWI Setting Subscribe for MWI Gof 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 f 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 56 STUN Setting MWI 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 you
33. lls you just pick up the phone to answer when ringing 88 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 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 or Bridge in accord with your network in WAN settings page 2 Select DHCP Client to automatically get an IP address from NAT router Remember to click the Submit button 4 Select Active ON in the SIP settings Service Domain page 5 Enter the Register Name Register Password Proxy Server and Outbound Proxy 6 Select ON in the STUN setting if Outbound Proxy is NOT available 89 7 Upon successful SIP registration the REG LED indicator will be ON and the LCD will show registered lt phone number gt 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 Dialling without will not dial out until the auto dial timer default 5 seconds
34. mand multi protocol virtual private networking over public networks such as the Internet This IP Phone has built in PPTP Client which allows connection to a PPTP based Virtual Private Network VPN such as VPN Broadband Router or IP PBX with PPTP function built in PPTP Settings You could set the PPTP server in this page PPTP GOf PPTP Server PPTP Usemame PPTP Password 50 PPTP Select On to enable PPTP function PPTP Server Enter PPTP Server s IP address or URL PPTP Enter login user name Username PPTP Enter password Password Application Diagram PPTP Server IP PBX VPN Router etc IP Phone Note This PPTP function is designed to connect to VOI 9300 which enables secured tunnel between the Phone and IP PBX 51 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 52 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
35. n the network when choosing CODECs 60 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 G726 16 ID 23 85 255 v 23 G726 24 ID 22 85 255 22 G726 32 ID 2 85 255 M 2 G726 40 ID 21 95 255 v 21 RFC 2833 ID 101 85 255 v 101 Two VoIP devices with different Codec ID will cause the interoperability issue If you are talking with others got some problems you may ask the other one what kind of Codec ID he use then you can change your Codec ID 61 Other Settings You can setup the Hold by RFC and QoS in this page To change these settings please follows your ITSP information The QoS 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 Gof Voice QoS Dif Sen 40 0 53 SIP QoS Dif Sen 40 0 53 SIP Expire Time 60 15 86400 sec Use DNS SRV Gof Submit Note For more information about these advanced features please ask your network administrator or service provider help desk 62 3 5 Others Auto Configuration function can be used to download the original co
36. nfigurations stored in the TFTP or FTP server Others Auto Config FXO Port MAC Clone Tones Advanced 63 Auto Config This feature allows service provider to provision their customer s IP Phone end to end By employing a TFTP FTP HTTP 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 Auto Configuration Setting You could enable disable the auto configuration setting in this page Auto Configuration Gor OTFTP OHTIP TFTP Server HTTP Server Exp 60 35 187 30 HTTP File Path Exp download FTP Server Exp 60 35 17 1 FTP Username FTP Password FTP File Path Exp file load Submit Note Auto Config is idea for ITSP or large network group to deploy VoIP devices easily 64 FXO Port The FXO Port is to configure and match the PSTN line impedance for each country This setting relates to your local telecom or Private Branch eXchange PBX system FXO Impedence Setting You could select the FXO impedence of the analog telephone by different country in this page FXO Port USA v Note FXO Port setting is for the VOI 7011 only 65 MAC Clone The MAC Clone function is to clone the MAC when only one MAC is available from ITSP Enable it to copy the MAC address of the Ethernet Card
37. o Dial Time i5 3 9 sec Use as send key 9 Yes ONo Use for IP dialing Yes ONo Auto dial Timer The inter digit timer Default is 5 seconds Use as send Enable or disable key as send key key Use for IP Enable or disable key as IP dialling key dialling 32 Flash Time Pressing quick on and off hook Flash allows you to use special features of your host PBX such as transferring an extension call or accessing optional telephone services such as Call Waiting The flash time depends on your telephone exchange or host PBX Flash Time Setting You could set the flash time in this page Flash Time 60 x10MS 9 120 Reset Note The Flash Time depends on your telephone exchange or Telephone Company Check system administrator for more information 33 Call Waiting You can enable the call waiting function in this page It allows answering another coming call by pressing flash key while holding the current call You may switch back to previous call by pressing flash key again Call Waiting Setting You could enable disable the call waiting setting in this page Call Waiting Oof Note Flash key means On hook and Off hook in short period without hanging up the call 34 Soft key You can configure the pickup and VMS key setting to co work with IP PBX in this page These keys are corresponding with Function keys VMS and Pick Up Soft key Setting You could configu
38. r ITSP for STUN information Message Waiting Indicator MWI in telephony is a Bellcore term for an FSK based telephone calling feature that illuminates an LED on select telephones to notify a telephone user of waiting voicemail messages on public telephone networks and PBXs When set to On a Subscribe for Message Waiting Indication will be sent periodically 57 Codec You can setup the Codec priority RTP packet length and VAD function in this page Codecs basically convert analog signals to digital form and vice versa Codec Settings You could set the codec settings in this page Codec Priority 1 2272 Codec Priority 3 v 227 ee Codec Priority 5 ies 6 BN 672524 v Codec Priority 7 G 726 32 72012 Codec Priority 9 GSM 0 711 amp 0 729 6 723 5 3K Sof Voice VAD 2 Voice VAD Sof 58 Codec Priority Adjust Codec priority to meet your requirement lower number shows higher priority RTP Packet Adjust Codec g711 g729 and g723 packet Length length G 723 5 3K Enables 5 3K bit s rate when use g723 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 VoIP call The higher t
39. rame A VID must be assigned for each VLAN 45 User Priority 802 1 P CFI Eight classes are defined by 802 1p Highest priority is seven which might go to network critical traffic such as Routing Information Protocol Values five and six might be for delay sensitive applications such as interactive video and voice CFI Canonical format indicator A 1 bit indicator that is always set to zero for Ethernet switches CFI is used for compatibility between Ethernet and Token Ring networks If a frame received at an Ethernet port has a CFI set to 1 then that frame should not be bridged to an untagged port Note The prioritization specification works at the media access control MAC framing layer of the OSI model To be compliant with 802 1p Layer 2 switches must be capable of grouping incoming LAN packets into separate traffic classes 46 DMZ In computer networks a DMZ demilitarized zone is a computer host or small network inserted as a neutral zone between a company s private network and the outside public network DMZ Setting You could configure your demilitarized zone setting in this page DMZ GOn DMZ Host IP 0 0 0 0 Enable the DMZ and enter the Host IP address into DMZ Host IP 47 Virtual Server The IP Phone can be configured as a virtual server This function is ideal for that remote users accessing Web or FTP services via the public IP address can be automatically redirecte
40. re the soft key setting in this page Pick up key Voice mail key IP Phone may pick up the incoming call for another IP Phone when registered in the same IP PBX When you hear other IP Phone is ringing you may pick up you phone and press Pick Up function key to answer for that IP Phone You may press the Speaker Phone key then Pick Up function key as well When registered in IP PBX with incoming voice message the LED VMS will start flashing To hear the message you may press the Speaker Phone key then VMS function key You may also pick up the handset and press VMS function key 35 Hot line When Hot Line mode is enabled you just lift up the handset and the IP Phone will call the Hot line number immediately The default for Hot Line mode is disabled Hot line Setting You could set the hot line in this page Use Hot Line Enable 9 Disable Hot line number Submit Note Hot Line Mode is very convenient for IP calling to Public Switching Telephone Network PSTN number through FXO Gateway 36 Alarm You can set the IP Phone as Alarm clock default is disabled Alarm Settings You could set the alarm time in this page Alarm OON OFF Alarm Time 0 20 hh mm Current time 2007 06 15 20 02 Reset IP Phone starts ringing at time you configured turn it off by press Speaker Phone or Off hook Note IP Phone rings different frequency while Alarm goes off 37 3 3 Network
41. rop Prefix No Replace Rule 1 002 8613 8662 Result a Pressing 8613xxx will result in dialling out 002 8613 xxx b Pressing 8662xxx will result in dialling out 002 8662 xxx Example 2 Drop Prefix Yes Replace Rule 2 006 002 003 004 005 007 009 Result a Pressing 002xxx will result in dialling out 006 xxx b Pressing 003xxx will result in dialling out 006 xxx Example 3 Drop Prefix No Replace Rule 3 009 12 Result a Pressing 12xxx will result in dialling out 009 12 xxx 30 Example 4 Drop Prefix No Replace Rule 4 007 5xxx 35xx 21 xx Result a Pressing 5xxx will result in dialling out 007 5 xxx b Pressing 534 will result in dialling out 534 Not matched c Pressing 35xx will result in dialling out 007 35 xxx d Pressing 356 will result in dialling out 356 Not matched e a Pressing 35668 will result in dialling out 35668 Not matched Example 5 Dial Now xx xXX 11X4XXXXXXXX 1 Pressing 00 01 02 99 will result in dialling out the same xx immediately 2 Pressing 00 01 02 99 will result in dialling out the same xx immediately 3 Pressing 110 111 119 will result in dialling out the same 11x immediately 4 Pressing 12345678 8 digits will result in dialling out 12345678 immediately This implies that the phone numbers with 9 or more digits are prohibited 31 Aut
42. tores up to 140 records VPN PPTP Client embedded One FXO port to forwarding calls Only VOI 7011 provides FXO port 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 12VDC 1A 5 Cable CD User Manual If any item is found missing or damaged please contact your local reseller for replacement 1 3 Optional DHM 1000 Lightweight Single Headset with Microphone 2 Hardware Description 2 1 LCD Display and Keypads The LCD display and keypads of IP Phone are as the following LCD Display X3 B mm 2 2 Front Panel VOI 7010 Headset RJ45 RJ45 VOI 7011 WAN LAN Headset RJ45 RJ45 Memory Card Use the memory card as a name index for speed dialler or extensions MENU ENTER 15 22 amp 27 6 T Br 9 9 REDIAL HOLD TRANSFER 2 3 Connection Diagram Optional Internet Headset 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 Cate
43. ty B While in conversation Party B may press Transfer key and should hear a dial tone Party B press Party C number and hang up to transfer to Party C Attendant Transfer 1 2 Party A calls Party B 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 Hang up from Party B and then Party A will transfer and connect to Party C 95 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 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 conversations 96 5 6 Direct IP to Direct IP Calling Applications The applications are for ADSL connection without NAT router as 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 Fi
44. ubnet as PC and NAT router 103 ADSL Modem H INTERNET NAT Routere WAN H 7 2928 Phone IP 192 168 1231 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 104
45. xed IP in the Network WAN settings page 2 Enter the items of IP Subnet Mask Gateway 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 97 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 Site 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 98 SIP Settings Ream ee Active G amp On Display Name 1636346
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