LevelOneメーカーVOI-7010の使用説明書/サービス説明書
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LevelOne VOI-7010 / VOI-701 1 SIP IP T elephone User Manual V er . 1.0 - 0707.
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iii Safety FCC W ARNING This equipment may generate or use radio frequency energy . Changes or modifications to this equipment may cause harmful i nterference unless the modifications are expressly approved in the instruction manual. The user could lose the authority to operate this equipment if an unaut horized change or modification is made.
iv T able of Content s 1. INTRODUCTION ............................................................. 1 1.1. F EA TURES .................................................................. 2 1.2. P ACKING C ONTENTS .......................................
v Network S t atus ............................................................... 39 W AN Settings ................................................................. 40 LAN Settings .................................................................. 42 DDNS Setting.
vi 5. SIP Settings......................................................... 84 6. NA T T ransversal .................................................. 86 7. Administrator ....................................................... 86 5. APPLICA TION EXAMPLE.
1 1. Introduction The VOI-7010 / VOI-701 1 IP Phone are an LCD V oIP Phone with SIP Protocols for V oice over IP (V oIP) applications. IP Phone can make a V oIP call over the ADSL Internet connection, and it provides one RJ45 WA N port for ADSL Internet connections plus one RJ45 LAN port for Notebook PC connection.
2 1.1. Features SIP v1 (RFC2543), v2 (RFC3261) with MD5 authentication (RFC2069 and RFC 2617) RJ45 x 2 for Ethernet W AN and LAN ports ITU-T G .
3 1.2. Packing Content s Open the shipping cartons of the Switch and c arefully unpacks its contents. The carton should contain the following items: ¡ SIP IP T el ephone ¡ Power Adaptor (12VDC/1A) ¡ Cat.5 Cable ¡ CD User Manual If any item is found missing or damaged, pleas e contact your local reseller for replacement 1.
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5 2. Hardware Description 2.1. LCD Display and Keyp ads The LCD display and keypads of IP Phone are as the following. Speed Dial Function Key LCD Display Menu Key V olume Up / Down Number Keypad s Han.
6 2.2. Front Panel VOI-7010 VOI-701 1 Power LAN RJ45 W AN RJ45 Headset Power LAN RJ45 W AN RJ45 Headset FXO.
7 Memory Card Use the memory card as a name index for speed dial ler or extensions..
8 2.3. Connection Diagram Note Public Switched T elephone Network (PSTN), which refers to the international telephone system based on copper wires carrying analog voic e data T elephone service carried by t he PSTN is often called plain old telephone service (POTS).
9 2.4. Inst allation 1. Connect IP Phone RJ45 W AN port to NA T 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.
10 registered in the SIP server . Note t hat # 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 wai t for answer . 2.5.
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12 3. W eb Configuration Y ou may enter the IP address from PC W eb browser to configure IP Phone. For example, enter http://192.168.1.100 from Web browser to display login page as follows.
13 System Information After login, you will see the system information like firmware version, Codec, etc in this page. Y ou may click the button list at the left hand side to configure the IP Phone.
14 3.1. Phone Book The Phone Book specifies pre-record phone list and speed dialling function, it a llows up to 140 records on the phone book..
15 Input the Position (0~139), Name and URL, then c lick the ¡Add Phone¡ button to enter . Note URL can be either complete strings or numbers only , it depends on your servic e provider . Example Phone Name URL Select 1 David 221 □ 2 Bill 221090@sipcall.
16 S peed Dial Setting For S peed Dial function you can add/delete S peed Dial number up to maximum 10 entries in S peed Dial Phone List..
17 If you need t o add a phone number into the S peed Dial list, you need to enter the position, the name, and the pho ne number (by URL type). When you finished a new phone lis t, 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.
18 3.2. Phone Setting The sub pages are as follows; Call Forward, SNTP , V olume, Melody (Ringer), DND, Auto Answer , Dial Plan, Flash T ime, Call W aiting, Soft-key , Hotline and Alarm functions.
19 Call Forward Y ou can have your incoming calls forwarded to a specified destination. Y ou can sele ct 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 .
20 All Fwd No S pecify All Forward number Busy Fwd No S pecify Busy Forward number No Answer Fwd No S pecify No Answer Forwa rd number No Answer Fwd Time Out S pecify the time period before forward calls Note Y ou have to set the T ime Out T imer to start to forward the calls.
21 SNTP Y ou can setup the primary and second SNTP Server IP Address, to get the date/time information. Y ou may also set the T ime Zone, and how long need to synchronize agai n. When you finished the setting, please click the ¡Submit¡ bu tton. SNTP (Simple Network Time Protocol) SNTP is an acronym that stands for Simple Network T ime Protocol.
22 V olume Raise or lower the sound level by using the V olume Control. For example, if it is difficult to hear the other party's voice; raise the Handset V olume, or If the other party has difficu lty hearing you; raise the Handset Gain level.
23 Handset V ol. Set the volume to hear from the handset Speaker V ol. Set the volume to hear from the S peaker Ringer V ol. Set the volume of ringer PSTN-Out V ol.
24 Ringer Y ou 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 se.
25 DND (Do Not Disturb) Y ou can setup the DND (Do Not Disturb) to keep the phone silence. Y ou may set this feature wh en you are in a meeting or busy .
26 Auto Answer Y ou may enable the Auto Answer function to answer the incoming call by FXO port. When the ring count ex ceeds the number set in Auto Answer Count er , the FXO port will auto answer and allow for extension calls from PSTN to V oIP and vice versa.
27 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 Counter Set time period before phone pick up the calls automatically PIN Code Enabled Enable the call restriction from PSTN line to V oIP or vice versa.
28 Dial Plan Dial plan and 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.
29 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.
30 Example 1 Drop 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 Dr.
31 Example 4 Drop Prefix No Replace Rule 4 007, 5xxx+35xx+21xx 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 .
32 Auto dial Timer The inter-digit timer . Default is 5 seconds Use # as send key Enable or disable ¡#¡ key as send key Use * for IP dialling Enable or disable ¡*¡ key as IP dialling key.
33 Flash T ime 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 telephon e services such as Call W aiting. The flash time depends on your telephone exchange or host PBX.
34 Call W aiting Y ou can enable the call waiting function in this page. It allows answering another coming call by pressing f lash key while holding the current call. Y ou 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.
35 Sof t-key Y ou can configure the pickup and VMS key setting to co-work with IP PBX in this p age. These keys are corresponding with Function keys [VMS] and [Pick Up] . IP Phone may pick up the incoming call for another IP Phone when registered in the same IP PBX.
36 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.
37 Alarm Y ou can set the IP Phone as Alarm clock, default is disabled. IP Phone st arts ringing at time you configured, turn it off by press [Speaker Phone] or Off-hook .
38 3.3. Network Y ou can check the Network status, and configure the W AN, LAN, DDNS, VLAN, DMZ, Virtual Server and PPTP settings in this section. Network S t atus W AN LAN DDNS VLAN DMZ Virtual Serve.
39 Network S t atus Y ou can check and show the current Network settings in this page. Interface 0 shows W AN port status, and Interface 1 shows LAN port status.
40 W AN Settings The W AN setting is used to configure the Ethernet port connects to the ADSL Modem/Router , or Ethernet switc h..
41 LAN Model T he default setting is NA T mode for IP Phone, and this enables the embedded NA T router between the LAN port and PC port. Y ou may change to Bridge Mode if you need NOT use the embedded NA T router . When setting to Bridge Mode, the W AN and the LAN ports will be bridged.
42 LAN Settings This embedded NA T is useful for ADSL users without NA T router , and it sep arates 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 .
43 DDNS Setting DDNS (Dynamic DNS) A serv ice that lets anyone on the Internet gain access to resources on your local network wh en the Internet address of that network is constantly changing. When it detect s that the IP address of the cable or DSL modem has changed, it notifies the DDNS service provider of the new address.
44 Y ou need to have a DDNS account before configuring the DDNS setting. Usually , most of the V oIP applications are working with a SIP Proxy Server . Nonetheless, you m ay have a DDNS account with a public IP address, and others can call you via the DDNS account.
45 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.
46 User Priority (802.1P) Eight classes are defined by 802.1p. Highest priority is seven , which might go to network-critical traffic such as Routing Information Protocol. V alues five and six might be for delay-sensitive applications such as interactive video and voice CFI CFI (Canonical format indicator).
47 DMZ In computer networks, a DMZ (demilitarized zone) is a computer host or small network inserted as a "neutr al zone" between a company's private network and the outside public network Enable the DMZ and enter the Host IP address into DMZ Host IP .
48 V irtual 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 redirected to local servers in the LAN.
49 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.
50 PPTP Point-to-Point T unnelling 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 virt ual private network (VPN) across TCP/IP-based data networks.
51 PPTP Select On to enable PPTP function PPTP Server Enter PPTP Server ¡ s IP address or URL PPTP Username Enter login user name PPTP Password Enter password Application Diagram Note This PPTP function is designed to connect to VOI-9300 which enables secured tunnel bet ween the Phone and IP PBX.
52 3.4. SIP Settings Y ou can setup the Service Domain, Port Settings, Codec Settings, R TP Setting, RPort Setting and Other Settings for SIP Proxy Server registrations in this page.
53 Underst anding the SIP SIP , the Session Initiation Protocol, is a signalling prot ocol for Internet conferencing, telephony , presence, events notification and instant messaging.
54 Service Domain Y ou may register up to three SIP accounts in the IP Phone. Y ou 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.
55 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.
56 DTMF Setting Y ou 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.
57 STUN Setting The STUN function must be enabled to work properly behind NA T when registered in SIP server . Y ou may enter the STUN server IP address and the STUN port number .
58 Codec Y ou can setup the Codec priority , RTP p acket length, and V AD function in this page. Codecs basically convert analog signals to digital form and vice versa.
59 Codec Priority Adjust Codec priority to meet your requirement, lower number shows higher priority . RTP Packet Length Adjust Codec g71 1, g729 and g723 packet length G .723 5.3K Enables 5.3K bit/s rate when use g723 V oice V AD V AD (V oice Activity Detection) is used to reduce the transmission rate during inactive speech periods.
60 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 s cheme also compresses PCM using advanced codebook technology .
61 Codec ID Y ou can setup the Codec ID in this page. Y ou need to follow the ITSP suggestion to setup these items. Note T wo V oIP 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.
62 Other Settings Y ou can setup the Hold by RFC and QoS in this page. T o 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.
63 3.5. Others Auto Configuration function can be used to downloa d the original configurations stored in the TFTP or FTP server . Others Auto Config FXO Port MAC Clone T ones Advanced.
64 Auto Config This feature allows service provider to provision their customer's IP Phone, end-to-end. By employing a T FTP / FTP / HTTP server , the provisioning server writes the configuration files needed to automatically configure the IP Phone.
65 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 Note FXO Port setting i.
66 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 installed by your ISP and replace the W AN MAC address with the MAC address of the IP Phone.
67 T ones The T o ne setting can be adjusted to generate Dial t one, Ring tone, Ring Back tone, and Busy tone for different countries Note T o meet your current system tone s ettings, please refer to PBX technical manual or ask telecom technician.
68 Advanced The advanced settings might be useful for s ome network requirements. 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.
69 Send Anonymous CID Select No if you subscribe to CallerID service on your PSTN line, otherwise Y es Management from W AN Select Y es to allow user manage the IP Phone from W AN Send Flash Event Select DTMF Event , the Flash will be sent as a DTMF event.
70 3.6. User Password Y ou may create the login name and password in this page. 3.7. Save Change Y ou must save the changes you have made, and click the Save button.
71 3.8. Up date User can update the IP Phone firmware when new firmware is available. Make sure no power off during the firmware upgrade. Up date New Firmware Auto Update Default Caution VOI-7010 and .
72 Up date Firmware 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) 2. Click the ¡Browse¡ button to choose the updated fi le location for HTTP download, or 3.
73 Caution VOI-7010 and VOI-701 1 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.
74 Auto Up date Settings The IP Phone provides three methods, TFTP , FTP or HTTP , to update new firmware as the following steps.
75 Note This function is mainly for your ISP settings only , ask your network administrator before change any parameters. Default Setting Y ou 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.
76 3.9. Reboot Y ou may click the Reboot button to restart, then IP Phone will automatically reboot with the stored configurations..
77 4. LCD Display and Keyp ad Y ou can use keypad to configure and to check the status of IP Phone. Make sure that the W AN port is connected to ADSL Ethernet, or you may hear a busy tone from the telep hone.
78 4.1. Keyp ad Descriptions Key Descriptions 1 ¡1¡, ¡-¡, ¡ ٫ ¡, ¡!¡, ¡?¡ 2 ¡2¡, ¡a¡, ¡b¡, ¡c¡, ¡A¡, ¡B¡, ¡C¡ 3 ¡3¡, ¡d¡, ¡e¡, ¡f¡, ¡D¡, ¡E¡, ¡F¡ 4 ¡4¡, ¡g¡,.
79 Key Name Descriptions VOL +/- This is for phone volum e settings. UP/DOWN Up↑ and Down↓ key s for LCD display . P . BOOK T o sho w the phone book list. CALL LOG T o show Incoming/outgoing calls history . DEL/MUTE T o del ete or to mute IP phone.
80 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.
81 3. Call setting 1 Call forward 1.All Forward: Activation: T o Enabled/Disabled this function. Number: Forward to a registered or URL Number . 2.Busy Forward. Activation: T o Enabled/Disabled this function. Number: Forward to a registered or URL Number .
82 5 V olume and Gain 1.Handset volume: Set Handset volume from 0~15 (max.) for you to hear . 2.S peaker volume: Set S peaker 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.
83 4. Network 1 W AN Setup 1 IP T ype: Fixed IP client DHCP client: PPPoE client: 2 Fixed IP setting: Host IP Subnet mask Gateway IP 3 PPPoE setting: User name Password 2 LAN Setup 1 Bridge 2 NA T 3 D.
84 5. SIP Settings Note T o 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.
85 2 Codec 1 Codec type G .71 1 uLaw: G .71 1 uLaw G .71 1 aLaw: G .71 1 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 V AD V oice Activity Detection Enable/Disable.
86 6. NA T T ransversal 1 STUN setting 1.STUN: STUN Enabled/Disabled 2.STUN server: Server I P 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.
87 5. Application Example Y ou 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 NA T enabled in IP Phone B.
88 5.1. PSTN Calling Applications: VOI-701 1 is default at the V oIP 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 t elephone. Configurations: The ¡ Auto Answer ¡ is OFF at default, and the function of extension call from SIP to PSTN is disabled.
89 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 NA T router .
90 7. Upon successful SIP registration, the REG LED indicator will be ON and the LCD will show registered <phone number>. Callings: 8. Pick up the phone, and you should hear a dial tone f or V oIP mode. 9. Press 1688# or 1688 to call the party with the registered SIP phone number 1688.
91 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 NA T router .
92 you must add the postfix ¡#¡. PIN Code is us ed 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.
93 5.4. PSTN-to-SIP Cal ling 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 NA T router . Configurations: 1. Same as in Example 2.
94 not ¡dodo¡ tone and the caller may press SIP number directly . 6. Press 1688# or 1688 to call the party with the registered SIP phone number 1688.
95 5.5. 3-W ay Co nference Calling Applications: The Call T ransfer and 3-W ay 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 T ransfer and Attendant T ransfer .
96 3-W ay Conference Call: 1. Party A calls Party B. 2. While in conversation, Party B may press Hold key to hold the call, and should hear a dial tone. 3. Party B calls Party C. 4. While in conversation, Party may press Conf. key to join in Party A for three-way conference.
97 5.6. Direct IP to Direct IP Calling Applications: The applications are for ADSL connection without NA T 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 .
98 5.7. FreeW orld Dialup (FWD) Applications: This shows how to use FWD as an example for f ree 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 instruct ions for registration.
99 SIP Settings Y ou 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 .
100 Codec Setting Callings: 1. Pick up the phone, and the LCD will show FWD ph one number <636346>. 2. 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 .
101 6. S pecification Model No. VOI-7010 V OI-701 1 Connector 1 x W AN 1 x LAN 1 x Headset Plug 1 x W AN 1 x LAN 1 x Headset Plug 1 x RJ1 1 FXO LCD Size 16 x 2 Network Protocol SIP v1 (RFC2543), v2(RF.
102 Codec G .71 1: 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 V AD & CNG to G .
103 7. T rouble Shooting 7.1. Do not hear dial tone? When you pick up the phone and hear a busy tone, it indicates the W AN port is NOT connected. The LCD will show Ethernet Error! Make sure the ADSL Ethernet cable is connected to the W AN port of IP Phone and Power Reset again.
104 Example: T o change IP PHONE IP address to the same subnet as PC and NA T 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 NA T DHCP server . 2. Press Menu_4_5 to read IP Addresses for W AN and LAN Ports, for example, 192.
デバイスLevelOne VOI-7010の購入後に（又は購入する前であっても）重要なポイントは、説明書をよく読むことです。その単純な理由はいくつかあります：
LevelOne VOI-7010をまだ購入していないなら、この製品の基本情報を理解する良い機会です。まずは上にある説明書の最初のページをご覧ください。そこにはLevelOne VOI-7010の技術情報の概要が記載されているはずです。デバイスがあなたのニーズを満たすかどうかは、ここで確認しましょう。LevelOne VOI-7010の取扱説明書の次のページをよく読むことにより、製品の全機能やその取り扱いに関する情報を知ることができます。LevelOne VOI-7010で得られた情報は、きっとあなたの購入の決断を手助けしてくれることでしょう。
LevelOne VOI-7010を既にお持ちだが、まだ読んでいない場合は、上記の理由によりそれを行うべきです。そうすることにより機能を適切に使用しているか、又はLevelOne VOI-7010の不適切な取り扱いによりその寿命を短くする危険を犯していないかどうかを知ることができます。
ですが、ユーザガイドが果たす重要な役割の一つは、LevelOne VOI-7010に関する問題の解決を支援することです。そこにはほとんどの場合、トラブルシューティング、すなわちLevelOne VOI-7010デバイスで最もよく起こりうる故障・不良とそれらの対処法についてのアドバイスを見つけることができるはずです。たとえ問題を解決できなかった場合でも、説明書にはカスタマー・サービスセンター又は最寄りのサービスセンターへの問い合わせ先等、次の対処法についての指示があるはずです。
