Page 1 LevelOne VOI-7010 / VOI-7011 SIP IP Telephone User Manual Ver. 1.0 - 0707...
Page 3: Ce Declaration Of Conformity
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 4: Table Of Contents
Table of Contents INTRODUCTION ............. 1 1.1..............2 EATURES 1.2............. 3 ACKING ONTENTS 1.3..............3 PTIONAL HARDWARE DESCRIPTION .......... 5 2.1. LCD D ........5 ISPLAY AND EYPADS 2.2.............. 6 RONT ANEL 2.3..........8 ONNECTION IAGRAM 2.4.
Page 5 Network Status............... 39 WAN Settings..............40 LAN Settings ..............42 DDNS Setting..............43 VLAN Settings..............45 DMZ ................47 Virtual Server ..............48 PPTP................50 3.4. SIP S ............52 ETTINGS Service Domain.............. 54 Codec................58 Codec ID ................ 61 Other Settings ..............
Page 6 SIP Settings............84 NAT Transversal..........86 Administrator ............86 APPLICATION EXAMPLE..........87 5.1. PSTN C ............88 ALLING 5.2. SIP- -SIP C ..........89 ALLING 5.3. SIP- -PSTN C ..........91 ALLING 5.4. PSTN- -SIP C ..........93 ALLING 5.5........
Page 7: Introduction
1. Introduction The VOI-7010 / VOI-7011 IP Phone are an LCD VoIP 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.
Page 8: Features
 Direct IP/URL Dial without SIP Proxy or Dial number via SIP server  Phone book stores up to 140 records  VPN PPTP Client embedded  One FXO port to forwarding calls * * Only VOI-7011 provides FXO port...
Page 9: 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 (12VDC/1A) ¡ Cat.5 Cable ¡ CD User Manual If any item is found missing or damaged, please contact your local reseller for replacement 1.3.
Page 11: 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 Menu Key Speed Dial Function Key REG / VMS Volume Up / Down Number Keypads Redial / Hold Transfer Handsfree Microphone...
Page 12: Front Panel
2.2. Front Panel VOI-7010 Headset Power RJ45 RJ45 VOI-7011 Headset Power RJ45 RJ45...
Page 13: Memory Card
Memory Card Use the memory card as a name index for speed dialler or extensions.
Page 14: Connection Diagram
2.3. Connection Diagram Power Optional Headset Internet PSTN 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 15: 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 16: Default Setting
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. In a moment, you should hear a ring back tone, and wait for answer.
Page 18: 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...
Page 19: 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-7010 and VOI-7011 use different firmware format, check it carefully before upgrade...
Page 20: 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 21 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 □ Jone 221080@192.168.12.234 □...
Page 22: 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 23 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 24: 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 Auto Answer Dial Plan Flash Time Call Waiting Soft-key...
Page 25: 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 26 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 27: 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. SNTP (Simple Network Time Protocol) SNTP is an acronym that stands for Simple Network Time Protocol.
Page 28: 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.
Page 29 Set the volume send out to the other side¡s handset Speaker Gain Set the volume send out to the other side¡s speaker PSTN-In Gain Set the volume send out to the other side¡s handset. Note PSTN function is only available on VOI-7011...
Page 30: Ringer
Ringer 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 31: Dnd (Do Not Disturb)
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 Always All incoming call will be blocked when enabled DND Period Set a time period and the phone will be blocked during the time period...
Page 32: 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 33 Enable the call restriction from PSTN line Enabled to VoIP or vice versa. PIN Code 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...
Page 34: Dial Plan
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. The auto dial timer specifies the elapse time between the dialling digits.
Page 35 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.
Page 36 Example 1 Drop Prefix 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 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...
Page 37 Example 4 Drop Prefix 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 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...
Page 38 Auto dial Timer The inter-digit timer. Default is 5 seconds Use # as send Enable or disable ¡#¡ key as send key Use * for IP Enable or disable ¡*¡ key as IP dialling key dialling...
Page 39: Flash Time
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.
Page 40: 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 41: Soft-Key
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]. IP Phone may pick up the incoming call for another IP Phone when registered in the same IP PBX.
Page 42: Hot Line
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. Note Hot-Line Mode is very convenient for IP calling to Public Switching Telephone Network (PSTN) number through FXO Gateway...
Page 43: Alarm
Alarm You can set the IP Phone as Alarm clock, default is disabled. 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.
Page 44: 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 45: 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 46: Wan Settings
WAN Settings The WAN setting is used to configure the Ethernet port connects to the ADSL Modem/Router, or Ethernet switch.
Page 47 LAN Model 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.
Page 48: Lan Settings
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.
Page 49: 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 50 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.
Page 51: Vlan Settings
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 Packets If you enable VLAN Packets and set the VID, User Priority, and CFI, then all the...
Page 52 User Priority Eight classes are defined by 802.1p. (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.
Page 53: 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 Enable the DMZ and enter the Host IP address into DMZ Host...
Page 54: Virtual Server
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 redirected 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.
Page 55 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. Application Diagram...
Page 56: Pptp
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-demand, multi-protocol, virtual private networking over public networks, such as the Internet.
Page 57 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 Note This PPTP function is designed to connect to VOI-9300 which enables secured tunnel between the Phone and IP PBX.
Page 58: 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 59 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 60: 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. Click ¡Active¡...
Page 61 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 62 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 63 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 64: 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 65 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 66 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.
Page 67: 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 68: 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 69: Others
3.5. Others Auto Configuration function can be used to download the original configurations stored in the TFTP or FTP server. Others Auto Config FXO Port MAC Clone Tones Advanced...
Page 70: Auto Config
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.
Page 71: Fxo Port
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 is for the VOI-7011 only...
Page 72: Mac Clone
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 WAN MAC address with the MAC address of the IP Phone. Note It is not recommended that you change the default MAC address unless required by your ISP...
Page 73: Tones
Tones The Tone setting can be adjusted to generate Dial tone, Ring tone, Ring Back tone, and Busy tone for different countries Note To meet your current system tone settings, please refer to PBX technical manual or ask telecom technician.
Page 74: Advanced
Advanced The advanced settings might be useful for some 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.
Page 75 Send Select No if you subscribe to CallerID Anonymous service on your PSTN line, otherwise Yes Management Select Yes to allow user manage the IP from WAN Phone from WAN Send Flash Select DTMF Event, the Flash will be sent Event as a DTMF event.
Page 76: User Password
3.6. User Password You may create the login name and password in this page. 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...
Page 77: 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...
Page 78: 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 79 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...
Page 80: Auto Update Settings
Auto Update Settings The IP Phone provides three methods, TFTP, FTP or HTTP, to update new firmware as the following steps...
Page 81: Default Setting
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.
Page 82: Reboot
3.9. Reboot You may click the Reboot button to restart, then IP Phone will automatically reboot with the stored configurations.
Page 83: 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 84: 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 85 Key Name Descriptions VOL +/- This is for phone volume settings. UP/DOWN Up↑ and Down↓ keys for LCD display. P. BOOK To show the phone book list. CALL LOG To show Incoming/outgoing calls history. DEL/MUTE To delete or to mute IP phone. CONF.
Page 86: 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 87: Call Setting
3. Call setting 1 Call forward 1.All Forward: Activation: To Enabled/Disabled this function. Number: Forward to a registered or URL Number. 2.Busy Forward. Activation: To Enabled/Disabled this function. Number: Forward to a registered or URL Number. 3.No Answer Forward. Activation: To Enabled/Disabled this function.
Page 88 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...
Page 89: Network
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 3 PPPoE setting: User name Password 2 LAN Setup 1 Bridge 2 NAT 3 DNS Server 1 Primary DNS 2 Secondary DNS 4 VLAN...
Page 90: 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 91 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 92: 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 93: 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 94: Pstn Calling
5.1. PSTN Calling Applications: VOI-7011 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 95: 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 96 Upon successful SIP registration, the LED indicator will be and the LCD will show registered <phone number>. Callings: Pick up the phone, and you should hear a dial tone for VoIP mode. Press 1688# or 1688 to call the party with the registered SIP phone number 1688.
Page 97: 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 98 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.
Page 99: 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 100 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.
Page 101: 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 102 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 103: 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 104: 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 105: 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 106: 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 107: Specification
6. Specification Model No. VOI-7010 VOI-7011 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 LCD Size 16 x 2 SIP v1 (RFC2543), v2(RFC3261)
Page 108 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 109: 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 110 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.

This manual is also suitable for:

Voi-7010

LevelOne VOI-7011 User Manual

Introduction

Hardware Description

Web Configuration

Lcd Display and Keypad

Application Example

Specification

Trouble Shooting

Quick Links

Need help?

Questions and answers

Related Manuals for LevelOne VOI-7011

Summary of Contents for LevelOne VOI-7011

This manual is also suitable for:

Table of Contents