LevelOne VOI-7000 User Manual
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LevelOne VOI-7000 User Manual

LevelOne VOI-7000 User Manual

Voip phone; poe voip phone
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LevelOne
VOI-7000 VoIP Phone
VOI-7100 PoE VoIP Phone
User Manual
Ver. 2.4 - 1008

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Summary of Contents for LevelOne VOI-7000

  • Page 1 LevelOne VOI-7000 VoIP Phone VOI-7100 PoE VoIP Phone User Manual Ver. 2.4 - 1008...
  • Page 2 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.

  • Page 3: Table Of Contents

    Table of Contents INTRODUCTION..................1 1.1....................2 EATURES 1.2.................. 3 ACKING ONTENTS HARDWARE DESCRIPTION ..............4 2.1. LCD D ..............4 ISPLAY AND EYPADS 2.2.................... 5 RONT ANEL 2.3................7 ONNECTION IAGRAM Without Broadband Router ................7 With Broadband Router..................
  • Page 4 Call Waiting ....................26 3.3....................27 ETWORK Network Status ....................28 Bridge ......................29 NAT Settings ....................30 DDNS Setting ....................32 3.4. SIP S ..................34 ETTINGS Service Domain ....................36 Codec ......................40 Codec ID ......................43 Other Settings ....................
  • Page 5 SIP Settings ..................58 NAT Transversal ................. 60 Administrator ..................60 APPLICATION EXAMPLE ............... 61 5.1. PSTN C ..................62 ALLING 5.2. SIP- -SIP C ................63 ALLING 5.3. SIP- -PSTN C ................65 ALLING 5.4. PSTN- -SIP C ................66 ALLING 5.5.

  • Page 6: Introduction

    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 configurations for the VoIP Phone. IP Phone can...

  • Page 7: Features

    1.1. Features  SIP v1 (RFC2543), 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 ...

  • Page 8: Packing Contents

    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 – Cat.5 Cable – CD User Manual If any item is found missing or damaged, please contact your local reseller for replacement...

  • Page 9: Hardware Description

    2. Hardware Description 2.1. LCD Display and Keypads The LCD display and keypads of IP Phone are as the following. LCD Display Handset VMS LED Menu Key Volume Up / Down Handsfree Number Keypads...

  • Page 10: Front Panel

    2.2. Front Panel VOI-7000 Power RJ45 RJ45 VOI-7100 Power RJ45 RJ45 (PoE)

  • Page 11: Memory Card

    Memory Card Use the memory card as a name index for speed dialler or extensions.

  • Page 12: Connection Diagram

    2.3. Connection Diagram Without Broadband Router Power Internet With Broadband Router Broadband Internet Router 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).

  • Page 13: Installation

    2.4. Installation Connect IP Phone RJ45 WAN port to NAT Router using a Category 5 LAN cable. Connect IP Phone RJ45 LAN port to Notebook PC using a Category 5 LAN cable. Connect DC power adaptor, and the LCD panel will start showing Loading Program! and System Initialized.

  • Page 14: Default Setting

    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.

  • Page 15: Web Configuration

    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. Enter the username and password into the blank field. The default settings are: Username: root Password: root Click the “Login”...

  • Page 16: 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. Caution VOI-7000 and VOI-7100 use different firmware format, check it carefully before upgrade...

  • Page 17: Phone Book

    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.
  • Page 18 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 Phone Name Select □ David □ Bill 221090@sipcall.org □...

  • Page 19: Speed Dial Setting

    Speed Dial Setting For Speed Dial function you can add/delete Speed Dial number up to maximum 10 entries in Speed Dial Phone List.
  • Page 20 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 “Delete Selected”...

  • Page 21: Phone Setting

    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...

  • Page 22: Call Forward

    Call Forward You can have your incoming calls forwarded to a specified destination. You can select the forward mode and enter the forward URL. 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.
  • Page 23 All Fwd No Specify All Forward number Busy Fwd No Specify Busy Forward number No Answer Fwd Specify No Answer Forward number 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.

  • Page 24: Sntp

    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.

  • Page 25: Volume

    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. Handset Vol.

  • Page 26: Melody

    Melody You may set ON the ringer and select different ringer type for Melody settings. Note Because the default ringer is ringer 1, it means the setting will remain as off if you switch On and select ringer 1...

  • Page 27: Block Setting

    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. 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”...

  • Page 28: Auto Answer

    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.
  • Page 29 Auto Answer Enable this function to answer the incoming calls from PSTN line automatically. It allows user to place call to Internet again. Auto Answer Set time period before phone pick up the Counter calls automatically PIN Code Enable the call restriction from PSTN line Enabled to VoIP or vice versa.

  • Page 30: Auto Dial

    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 Time The inter-digit timer. Default is 5 seconds...

  • Page 31: Call Waiting

    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. Note Flash key means On-hook and Off-hook in short period without hanging up the call.

  • Page 32: Network

    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 DDNS VLAN Virtual Server PPTP...

  • Page 33: Network Status

    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.

  • Page 34: Bridge

    Bridge The Bridge setting is used to configure the Ethernet port connects to the ADSL Modem/Router, or Ethernet switch.

  • Page 35: Nat Settings

    Bridge When setting 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.

  • Page 37: Ddns Setting

    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.
  • Page 38 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 In this example, the other user can place VoIP calls to your IP Phone directly by your domain address.

  • Page 39: Sip Settings

    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...
  • Page 40 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.

  • Page 41: Service Domain

    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.
  • Page 42 Realm (1 ~ 3) Active Enable the SIP account Display Name Enter the name you want to display User Name Enter the User Name given by your ITSP Register Name Enter the Register Name given by your ITSP Register Enter the Register Password given by Password your ITSP Domain Server...
  • Page 43 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.
  • Page 44 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.

  • Page 45: Codec

    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.
  • Page 46 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.
  • Page 47 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.

  • Page 48: Codec Id

    Codec ID You can setup the Codec ID in this page. You need to follow the ITSP suggestion to setup these items. Note 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.

  • Page 49: Other Settings

    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.

  • Page 50: Auto Config

    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.
  • Page 51 Note Auto Config is idea for ITSP or large network group to deploy VoIP devices easily...

  • Page 52: Icmp Setting

    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 Not Echo ICMP is used to acknowledge and echo for the Ping request. IP Phone will echo for the IP Ping request at default.

  • Page 53: Save Change

    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 Default Caution VOI-7000 and VOI-7100 use different firmware format, check it carefully before upgrade...

  • Page 54: Update Firmware

    Update Firmware The IP Phone provides two methods, HTTP or TFTP, to update new firmware as the following steps: 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”...
  • Page 55 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...

  • Page 56: Default Setting

    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. 3.9. Reboot You may click the Reboot button to restart, then IP Phone will automatically reboot with the stored configurations.

  • Page 57: Lcd Display And Keypad

    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.

  • Page 58: Keypad Descriptions

    4.1. Keypad Descriptions Descriptions “1”, “-“, “٫”, “!”, “?” “2”, “a”, “b”, “c”, “A”, “B”, “C” “3”, “d”, “e”, ”f”, “D”, “E”, “F” “4”, “g”, “h”, “I”, “G”, “H”, “I” “5”, “j”, “k”, “l”, “J”, “K”, “L” “6”, “m”, “n”, “o”, “M”, “N”, “O” “7”, “p”, “q”, “r”, “s”, “P”, “Q”, “R”, „S”...
  • Page 59 Key Name Descriptions PHONEBOOK To show the phone book list. To show Incoming calls history. To show Outgoing calls history. The “Enter” is for setting selections. ENTER The “Menu” key is to set the IP Phone MENU ▲▼◄► Use navigate keys to select menu items. VOL+/- Set the volume High/Low “Transfer”...

  • Page 60: Lcd Menu

    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.
  • Page 61 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...

  • Page 62: Network

    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 IP client DHCP client: PPPoE client: 2 Fixed IP setting: Host IP Subnet mask Gateway IP 3 PPPoE setting: User name Password...

  • Page 63: Sip Settings

    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:...
  • Page 64 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 2 VAD Voice Activity Detection Enable/Disable. 3 RTP Setting 1 Outband DTMF: Outband DTMF Enabled/Disabled 2 Duplicate RTP No duplicate:...

  • Page 65: Nat Transversal

    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.

  • Page 66: Application Example

    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 B. ADSL Connections with external NAT Router...

  • Page 67: Pstn Calling

    5.1. PSTN Calling Applications: VOI-7100 is default at the VoIP mode. For PSTN calls, you may just pick up the phone, press 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.

  • Page 68: Sip-To-Sip Calling

    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 A or B. Both parties are registered to SIP server under NAT router.
  • Page 69 Dialling without # will not dial out until the auto dial timer (default=5 seconds) elapsed.

  • Page 70: Sip-To-Pstn Calling

    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 A and B. Both parties are registered to SIP server with either fixed real IP or private IP under NAT router.

  • Page 71: Pstn-To-Sip Calling

    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: Same as in Example 2. Select “ON”...

  • Page 72: 3-Way Conference Calling

    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.
  • Page 73 3-Way Conference Call: Party A calls Party B. 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.

  • Page 74: Direct Ip To Direct Ip Calling

    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.

  • Page 75: Freeworld Dialup (Fwd)

    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.

  • Page 76: Sip Settings

    SIP Settings 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 number>;...

  • Page 77: Codec Setting

    Codec Setting Callings: Pick up the phone, and the LCD will show FWD phone number <636346>. 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.

  • Page 78: Specification

    6. Specification Model No. VOI-7000 VOI-7100 1 x WAN 1 x WAN 1 x LAN 1 x LAN Connector 1 x Headset Plug 1 x Headset Plug 1 x RJ11 FXO 16 x 2 LCD Size SIP v1 (RFC2543), v2(RFC3261)
  • Page 79 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 & CNG to G.729 VAD: Voice activity detection CNG: Comfortable noise generator Voice LEC: Line echo canceller Packet Loss Compensation Adaptive Jitter Buffer...

  • Page 80: Trouble Shooting

    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.
  • Page 81 Example: To change IP PHONE IP address to the same subnet as PC and NAT router 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.

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Voi-7100

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