LevelOne VOI-7010 / VOI-7011 SIP IP Telephone User Manual Ver. 1.
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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.
Table of Contents 1. INTRODUCTION ............................................................. 1 1.1. 1.2. 1.3. 2. HARDWARE DESCRIPTION .......................................... 5 2.1. 2.2. 2.3. 2.4. 2.5. 2.6. 3. FEATURES .................................................................. 2 PACKING CONTENTS.................................................... 3 OPTIONAL ................................................................... 3 LCD DISPLAY AND KEYPADS .......................................
Network Status............................................................... 39 WAN Settings................................................................. 40 LAN Settings .................................................................. 42 DDNS Setting................................................................. 43 VLAN Settings................................................................ 45 DMZ ............................................................................... 47 Virtual Server ...
5. 6. 7. 5. SIP Settings......................................................... 84 NAT Transversal.................................................. 86 Administrator ....................................................... 86 APPLICATION EXAMPLE............................................. 87 5.1. PSTN CALLING ......................................................... 88 5.2. SIP-TO-SIP CALLING ................................................ 89 5.3. SIP-TO-PSTN CALLING....................................
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. With the embedded NAT/DHCP server, IP Phone can be easily configured for different network diagrams by PC Web browser and telephone keypads.
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.
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.
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2. Hardware Description 2.1. LCD Display and Keypads The LCD display and keypads of IP Phone are as the following.
2.2.
Memory Card Use the memory card as a name index for speed dialler or extensions.
2.3. Connection Diagram Power Optional Headset WAN Internet LAN 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).
2.4. Installation 1. Connect IP Phone RJ45 WAN port to NAT Router using a Category 5 LAN cable. 2. Connect IP Phone RJ45 LAN port to Notebook PC using a Category 5 LAN cable. 3. Connect DC power adaptor, and the LCD panel will start showing Loading Program! 4. and System Initialized. The LCD panel will show Date, Time and No service without SIP registration, or after successful SIP registration. 5. Pick up the phone, and the LCD panel will show IP Dialling..
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. 2.5. Default Setting IP Address : 192.168.1.100 (LAN) Login Name : root Password : root 2.6. Reset to Default Press MENU / 7.Administrator / 2.Default setting / 1.
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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¡ button will enter the management information page for system setup.
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.
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.
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 URL Select 1 David 221 □ 2 Bill 221090@sipcall.org □ 3 Jone 221080@192.168.12.
Speed Dial Setting For Speed Dial function you can add/delete Speed Dial number up to maximum 10 entries in Speed Dial Phone List.
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¡ 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.
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.
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.
All Fwd No Specify All Forward number Busy Fwd No Specify Busy Forward number No Answer Fwd No Specify No Answer Forward number No Answer Fwd Time Out Specify the time period before forward calls Note You have to set the Time Out Timer to start to forward the calls.
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.
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. Set the volume to hear from the handset Speaker Vol. Set the volume to hear from the Speaker Ringer Vol. Set the volume of ringer PSTN-Out Vol. Set the PSTN volume for you to hear Handset Gain 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.
Ringer You may set ON the ringer and select different ringer type for Melody settings.
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 When the time in ¡From¡ is greater than ¡To¡, the Block time will be from Day 1 to Day 2.
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.
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 VoIP or vice versa. PIN Code Set the PIN code. User requires to enter correct code which correspond with before get second dial tone.
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.
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.
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 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+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 Dial Now: *xx+#xx+11x+xxxxxxxx 1) Pressing *00, *01, *02, ..., *99 will result in dialling out the same *xx immediately.
Auto dial Timer The inter-digit timer.
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. Note The Flash Time depends on your telephone exchange or Telephone Company.
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.
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. 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.
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.
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.
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 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.
WAN Settings The WAN setting is used to configure the Ethernet port connects to the ADSL Modem/Router, or Ethernet switch.
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. IP Type There are three selections for WAN: Fixed IP, DHCP Client, and PPPoE modes. This WAN setting is for the LAN port when set in NAT mode. The WAN default is at DHCP Client Mode.
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.
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.
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.
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 incoming packets will be checked with the IP Address and the VID. VID (802.
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. Values five and six might be for delay-sensitive applications such as interactive video and voice CFI 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.
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 IP.
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.
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.
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.
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 between the Phone and IP PBX.
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.
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.
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.
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 Password Enter the Register Password given by your ITSP Domain Server Enter the Domain Server given by your ITSP Proxy Server Enter the Proxy Server given by your ITSP Outbound Proxy Enter the Outbound Proxy of ITSP.
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.
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.
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 Priority Adjust Codec priority to meet your requirement, lower number shows higher priority. RTP Packet Length Adjust Codec g711, g729 and g723 packet 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.
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.
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.
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.
3.5. Others Auto Configuration function can be used to download the original configurations stored in the TFTP or FTP server.
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.
FXO Port The FXO Port is to configure and match the PSTN line impedance for each country.
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.
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.
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. 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.
Send Anonymous CID Select No if you subscribe to CallerID service on your PSTN line, otherwise Yes Management from WAN Select Yes to allow user manage the IP Phone from WAN Send Flash Event Select DTMF Event, the Flash will be sent as a DTMF event. Select SIP INFO, Flash is transmitted by SIP INFO messages SIP Encrypt This feature only work with ITSP required. PPPoE Retry Period Set Re-connect time period when DSL PPPoE connection is disconnected.
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.
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 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 file location for HTTP download, or 3. 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.
Auto Update Settings The IP Phone provides three methods, TFTP, FTP or HTTP, to update new firmware as the following steps 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.
3.9. Reboot You may click the Reboot button to restart, then IP Phone will automatically reboot with the stored configurations.
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.
4.1. Keypad Descriptions Key Descriptions 1 ¡1¡, ¡-¡, ¡٫¡, ¡!¡, ¡?¡ 2 ¡2¡, ¡a¡, ¡b¡, ¡c¡, ¡A¡, ¡B¡, ¡C¡ 3 ¡3¡, ¡d¡, ¡e¡, ¡f¡, ¡D¡, ¡E¡, ¡F¡ 4 ¡4¡, ¡g¡, ¡h¡, ¡I¡, ¡G¡, ¡H¡, ¡I¡ 5 ¡5¡, ¡j¡, ¡k¡, ¡l¡, ¡J¡, ¡K¡, ¡L¡ 6 ¡6¡, ¡m¡, ¡n¡, ¡o¡, ¡M¡, ¡N¡, ¡O¡ 7 ¡7¡, ¡p¡, ¡q¡, ¡r¡, ¡s¡, ¡P¡, ¡Q¡, ¡R¡, ¡S¡ 8 ¡8¡, ¡t¡, ¡u¡, ¡v¡, ¡T¡, ¡U¡, ¡V¡ 9 ¡9¡, ¡w¡, ¡x¡, ¡y¡, ¡z¡, ¡W¡, ¡X¡, ¡Y¡, ¡Z¡ 0 ¡0¡, ¡space¡ * ¡*¡, ¡¡¡, ¡:¡, ¡@¡ # Start dialling process.
Key Name VOL +/UP/DOWN P. BOOK Descriptions This is for phone volume settings. Up↑ and Down↓ keys for LCD display. To show the phone book list. CALL LOG To show Incoming/outgoing calls history. DEL/MUTE To delete or to mute IP phone. CONF. For 3-Way Conference Calls. M1~M4 These are for 4 speed dial numbers. LINE1~LINE3 DND For Realm 1 to Realm 3 SIP registrations. To enable/disable DND for call reject. FORWARD This is for forward function. VMS This is for voice messages.
4.2. LCD Menu 1. Phone Book 1.Search Search Phone Book 2.Add entry Add new phone number to phone book 3.Speed dial Add speed dial phone number 4.Erase all Erase all phone number 2. Call History 1.Incoming calls Show all incoming call. 2.Dialed numbers Show all dialled call. 3.Erase record Delete call history. 1 All: Delete all call history. 2 Incoming: Delete all incoming call. 3 Dialled: Delete all dialled out call.
3. Call setting 1 Call forward 1.All Forward: Activation: Number: To Enabled/Disabled this function. Forward to a registered or URL Number. 2.Busy Forward. Activation: Number: To Enabled/Disabled this function. Forward to a registered or URL Number. 3.No Answer Forward. Activation: Number: To Enabled/Disabled this function. 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: 2.By Period: 3.
5 Volume and Gain 1.Handset volume: 2.Speaker volume: 3.Handset Gain: 4.Speaker Gain: Set Handset volume from 0~15 (max.) for you to hear. Set Speaker phone volume from 0~15 (max.) for you to hear. Set Handset Gain from 0~15 (max.) for remote site to hear. Set Speakerphone Gain from 0~15 (max.) for remote site to hear. 6 Ringer 1.Ringer volume: Ringer volume selection from 0~15 (max.). Ringer tone selection from 1~4. 2.Ringer type: 7 Auto Dial Auto Dial time selection from 3~9 seconds.
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 1 Activation 2 VID 3 Priority 4 CFI 5 Status: Show IP addresses of WAN, LAN and MAC address (use UP/Down keys).
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.
2 Codec 1 Codec type G.711 uLaw: G.711 aLaw: G.723: G.729: G.726-16: G.726-24: G.726-32: G.726-40: 2 VAD G.711 uLaw G.711 aLaw G.723.1 G.729A G.726 16Kbps G.726 24Kbps G.726 32Kbps G.726 40Kbps Voice Activity Detection Enable/Disable. 3 RTP Setting 1 Outband DTMF: 2 Duplicate RTP Outband DTMF Enabled/Disabled No duplicate: One duplicate: Two duplicate: 4 RPort Setting No resend voice packets. Resend voice packets once. Resend voice packets twice. RPort Enabled/Disabled.
6. NAT Transversal 1 STUN setting 1.STUN: 2.STUN server: STUN Enabled/Disabled Server IP Address 7. Administrator 1 Auto Config 1 Config Mode: Select Disable/TFTP/FTP/HTTP for auto config function with server. 2 TFTP server: Set the TFTP server IP address. 3 FTP server: Set the FTP server IP address. 4 FTP Login Name: Set the login name to the FTP server. 5 FTP Password: Set the Password to the FTP server.
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.
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.
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. 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.
7. Upon successful SIP registration, the REG LED indicator will be ON and the LCD will show registered . 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) elapsed.
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. Configurations: 1. Same as in Example 2. 2. Select ¡ON¡ in the ¡SIP settings / STUN setting¡ page, if Outbound Proxy is NOT available. 3.
you must add the postfix ¡#¡. prevent from call piracy. call disconnect. PIN Code is used to Incorrect PIN Code will result in If PIN code is OFF, the caller may press PSTN number directly. 7. Press 7654321 to call the PSTN party number of 7654321.
5.4. PSTN-to-SIP Calling Applications: The applications can be for ADSL connections as in both Diagrams A and B. Both parties are registered to SIP server with either fixed real IP or private IP under NAT router. Configurations: 1. Same as in Example 2. 2. Select ¡ON¡ in the ¡SIP settings / STUN setting¡ page, if Outbound Proxy is NOT available. 3. Select ¡ON¡ for the ¡Auto Answer¡ and ¡PIN Code¡ in Call settings. Set the Auto Answer Ring Counter, e.g. 3, and the PIN code, e.g. 1234. 4.
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. In a moment, you should hear a ring back tone, and wait for the VoIP called party to answer.
5.5. 3-Way Conference Calling Applications: The Call Transfer and 3-Way Conference Call applications are for calls among Parties A, B, and C. Three parties are registered to SIP server with either fixed real IP or private IP. There are two kinds of call transfer; Blind Transfer and Attendant Transfer. Blind Transfer: 1. Party A calls Party B. 2. While in conversation, Party B may press Transfer key, and should hear a dial tone. 3. Party B press [Party C number] # and hang up to transfer to Party C.
3-Way 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. 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.
5.6. Direct IP to Direct IP Calling Applications: The applications are for ADSL connection without NAT router as in Diagram A. Both parties are with fixed real IP. The Direct IP calling works when both calling and answering parties are with known fixed IP. SIP server registrations are not required in this application. Configurations: 1. Select ¡Fixed IP¡ in the ¡Network / WAN settings¡ page, 2. Enter the items of IP, Subnet Mask, Gateway IP, 3. Click the ¡Submit¡ button. Callings: 4.
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.
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. automatically. The IP Phone will reboot After boot up, the SIP setting page will show ¡Registered¡, and the LCD will show registered ; it will shows ¡No service¡ otherwise.
Codec Setting Callings: 1. Pick up the phone, and the LCD will show FWD phone 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.
6. Specification Model No.
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 In-Band DTMF DTMF Function Out-of Band DTMF SIP Info QoS ToS field NAT Traversal STUN Web Browser Configuration Keypad Power DC12V, 1.
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.
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. 2. Press Menu_4_5 to read IP Addresses for WAN and LAN Ports, for example, 192.168.62.51. 3. Enter from IE web browser http://192.168.62.51 to login IP PHONE web page for configurations.
