VOI-8001 VOI-8002 VOI-8003 8-Port H.323/SIP VoIP Gateway User Manual 1 Ver. 1.
Table of contents CHAPTER 1. INTRODUCTION ...............................................................3 1.1 OVERVIEW .....................................................................................................................................................3 1.2 PACKAGE CONTENTS .....................................................................................................................................3 1.3 KEY FEATURE.....................................................................
Chapter 1. Introduction 1.1 Overview The VOI-800x series VoIP Gateway is equipped with 8 standard phone ports, one 10/100BaseTX Fast Ethernet WAN port, and one 10/100BaseTX Fast Ethernet LAN ports. With the integration of both voice and data, the offers ability to route data information into network solution VoIP Gateway has voice support that includes Quality of Service (QoS), voice compression, echo cancellation, dynamic latency (jitter) buffers, silence suppression, and comfort noise generation.
1.3 Key Feature VoIP Support 8 simultaneous VoIP calls Support T.38 FAX relay Support QoS(ToS) for VoIP Compliant with H.323 / SIP VoIP standard protocol Extensible by external IVR/CDR/Billing servers for value- added application Support register up to 4 Gatekeepers / Proxy servers Support worldwide off net call by ITSP service Support Multiple dialing plan / Call hunting group Adaptive Jitter Buffer function Multiple call profile for adjust VAD, Audio CODEC, H.
Chapter 2 Getting Started 2.1 Front Panel VOI-8001 VOI-8002 VOI-8003 (Ready) Flash: It means when the VoIP Gateway registration fail. (Ready) Constant: It means when the VoIP Gateway registration successes. (Status): This LED will flash quickly when the VoIP Gateway is either performing a self test of booting up. (Power): The LED light on when power on. (WAN): The LED light on when WAN port was connected, flash quick when WAN port transmission Voice or data.
2.2 Rear Panel VOI-8001 VOI-8002 VOI-8003 DC12V: For the included power adapter. Be sure to use only the 12VDC/2A power adapter included with the product. Using the wrong power adapter can damage the product and void the warranty. Reset: Clear all settings and restore them to the initial values present when the device was purchased. After performing the reset, make sure to redefine the IP settings for the device in the ‘Connection’.
Chapter 3 Configuration The default setting of DHCP Server inside VoIP Gateway is turn ON, So please set up your PC TCP/IP network as “Get IP Automatically” from DHCP to get internal IP from VoIP Gateway. By default, The VoIP Gateway will become the network gateway and default IP is 192.168.22.1 and will assign your PC IP as 192.168.22.X. Please go to “Control Panel”→”Network”.
3.1 Connection Click “Connected Type” option below “System Setup\Wan” item: Please select the type of Internet connection you have and set up the VoIP Gateway to use the Dynamic IP Address, Static IP Address, PPPoE, PPTP or L2TP connection. If your ISP has not given you an IP address, select Dynamic IP Address (default). If you have been given a specific IP address, select Specify an IP Address. To use Static-IP ADSL connection, please select “Static IP Address” and enter WAN IP settings.
To use PPPoE ADSL connection, please select Yes in use “PPPoE” ADSL service and enter the “username” and “password” in the PPPoE setup section. Most of ADSL ISPs assign dynamic-IP settings to the VoIP Gateway when using PPPoE. Please select Obtain IP Address Automatically(default setting). You can leave the Primary and Secondary DNS IP settings in default. The VoIP Gateway automatically obtains these settings from your ISP when the PPPoE connection is successfully established.
3.2 System status This page reveals the status of the VoIP Gateway including WAN, LAN and some hardware/firmware information.
3.3 Register Server If this VoIP Gateway wants to use SIP Proxy or GateKeeper service to transfer the VoIP call, you can input the server information here. The VoIP Gateway can register to up to four servers simultaneously. This page reveals the status of the server registration information. Here is the server configuration page, please ask the information form your ITSP. Remark: For Notify remark for this rule. Please use UNDERLINE to replace the SPACE due to HTTP protocol limitation.
3.4 VoIP Call Out User key in the phone number through phone set dial pad, then VoIP Gateway translate the phone number by the routing table setting here to destination IP & dial out number then Call out via network protocol Remark: For Notify remark for this rule. Please use UNDERLINE to replace the SPACE due to HTTP protocol limitation. Area Code: Define the Prefix number fit this rule, any phone number prefix digits matched with the rule will call out by this rule define.
IP address, for example: 168.56.9.22 URL, route via URL. For example: www.inphonex.com .This VoIP Gateway can setup to register to DDNS service (/System Setup /Advanced/ Dynamic DNS/) to let user call out to another VoIP Gateway with dynamic IP by URL. rsn , route via server, it will get the destination IP by server setting (/VoIP Setup/Register server/) in advance. For example: rs1 for server 1. rs2 for server 2. rs for all the server available ( search sequence: rs1 > rs2 > rs3 > rs4).
Chapter 4 Web UI Management 4.1 Access to Web UI The VoIP Gateway provide user friendly Web interface to let you configure your VoIP Gateway function The default setting of DHCP Server inside VoIP Gateway is turn ON, So please set up your PC TCP/IP network as “Get IP Automatically” from DHCP to get internal IP from VoIP Gateway. By default, The VoIP Gateway will become the network gateway and default IP is 192.168.22.1 and will assign your PC IP as 192.168.22.X. Please go to “Control Panel”→”Network”.
4.2 Web UI Management The VoIP Gateway provide user friendly Web interface to let you configure your VoIP Gateway function and VoIP function. There are a help on line content within each setting page. Please press Help hyperlink to view the on line help. There are 3 main functions for web, VoIP, System Setup (VoIP Gateway) & System maintenance. Each function is setup by the function below: 4.2.
System Maintenance function Configurations Backup/Restore VoIP Module Backup/Restore Reboot System Save Modification Gateway Manual overview 1
4.2.2 VoIP Function 4.4.2.1 VoIP Setup/ Port Status/ This page will display the current and last time VoIP call status & result. a. The PC time : will show the date & time that your connected PC now. b. The VoIP Gateway time : will show the date & time on this VoIP Gateway, the date& time may get from SNTP server or setting from your PC. You may set the SNTP server from /System Setup/Administrator/Date & Time/. A. Ports Message a. Port: display the port number, e.g. 1 or 2. b.
Out: There is a phone call made from Network VoIP and pick up by phone set. e. Connected IP: The remotely party IP of this VoIP call. f. Caller ID: Caller ID received from telephone line port. g. Start Time: Date & time of this VoIP call begin on this port. h. End Time: Date & Time of last VoIP call End on this port. i. Talking Sec: Total talked seconds of last VoIP call on this port. j. Dialed number: On the VoIP call out (line status display “In”).
4.2.2.2 VoIP/Line Configure/ Line Setting /VoIP Setup/Line Configure/Line Setting/ This page will setup the phone line information each port. a. Port: display the port number, e.g. 1 or 2. b. Interface: Telephone interface type: a. FXO: for connect to telephone line or PBX extension line. FXS: for connect to regulate phone set Name: Line name for this port. This will send and display on the remote side during VoIP call b. Line number: Telephone number assigned to this line. c.
when you off hook the phone connected to the line 1, it will directly call to another VoIP gateway location at 168.56.09.22 and dial 601. You can enable the line 1 as hot line, and add a routing rule on the routing table on /VoIP Setup/Routing Setup/VoIP Call Out/ to assign the AREA CODE to hl1 to handle the VoIP Gateway rule for hot line function. And please also remember to Strip 3 digits to stripe the “hl1” symbol and remember add real phone number you want to dial on Prefix.
4.2.2.3 Line configure/ Tone Setting /VoIP Setup/Line Configure/ Tone Setting A.Call Progress Tone This page defines the tones generated to the phone connected to the phone port. The cadence of CPT is been defined here also. All lines use same tone parameters. After modify the tone parameters, you must save modify then 26H26H26H 27H27H 27H Reboot to let the modified parameters work. Detect Voice Busy Cycle: Use the parameters to automatic detect cadence busy tone.
Busy: Define the busy tone for generate. Ring: Define the ring back tone for generate b. Low freq: Lower frequency for defined tone c. High freq: Higher frequency for defined tone. Each tone can define two frequencies, if only one frequency needed, please leave High Frequency to 0. d. T_ON_1, T_OFF_1, T_ON_2, T_OFF_2: The cadence pattern of up to four intervals for each dual-frequency. Minimum Cadence value is 30msec.
4.2.2.4 Line configure/ Line Feature /VoIP Setup/Line Configure/ Line Feature This page defines the feature on the phone port of the VoIP Gateway. A. Dial Pause signal length(as ,)[100~3000] ms: Define the pause time (ms) of the “,” on the /Routing Setting/VoIP Call Out/. This pause time is usually for time delay when connect to PBX and used for seize the CO line. The default pause time is 1000ms. The input range is between 100 to 3000 ms. User can use more then one “,” to get longer delay time. B.
Current Drop-> 3 S: Enable FXS interface to generate three seconds Current Drop signal. C. Called Number Relay on FXS ： Define when use the FXS interface to outbound call, resend or Drop out the dialed number. Drop out: Do not send the dialed number. When use the FXS port direct connect to phone set for outbound call, please enable the “Drop out” function to avoid hear the unnecessary dialed number when answer the phone call. Resend: Resend the dialed number.
Line number Priority: The 1st line has high priority; it will always ring the 1st line if it is available. When 1st line is busy, it will try to ring 2nd line if it is free. Rotation: 1st line ring first, then 2nd line ring next time, when the latest line ring this time, it will come back to ring 1st line next time. All: Ring all phone lines if it is available. Sequence: Ring all the available phone line one by one, the ring period for ring each phone is definable. Period (sec.
4.2.2.5 Line configure/ Line Polarity /VoIP Setup/Line Configure/ Line Polarity This page defines the Polarity on the phone port of the VoIP Gateway. If use the normal phone set to connect gateway, please select “Normal”. If use PBX or special PSTN line (support polarity invert), then please select “Invert”. Please remember to press the Modify button to take it effect. For store back to flash memory, please press Save Modification (/System Maintenance/Save Modification/).
4.2.2.6 Routing Setup/ VoIP Call Out Setting /VoIP Setup/Routing Setup/VoIP Call Out This page let you define the routing rule for Call out to VoIP. (User press the phone number through phone set dial pad, then VoIP Gateway translate the phone number by the routing table setting here to destination IP & dial out number then Call out via network protocol).Here can define some special keyword like IPIVR, PSTN as destination for some special function also.
translate for call out rule without needed to press # key. b. FirstDigitTime: Define the waiting time (seconds) for user press phone number first digit. User need to press first digits before the setting time (seconds) defined here, if VoIP Gateway wait for the defined seconds and there is no any digits press, the VoIP Gateway will stop to wait and feedback the user busy tone. c. OtherDigitTime: Define the waiting time (seconds) for user press phone number secondary & the rest digits.
b. Remark: Remark for this routing rule. Please use UNDERLINE to replace the SPACE due to HTTP protocol limitation. c. Area Code: Define the Prefix number fit this rule, any phone number prefix digits matched with the rule will call out by this rule define. Please Notify there is a compare order rule on this routing table.
for gatekeeper 1. gk2 for gatekeeper 2. gk for all the gatekeepers available ( search sequence: gk1 > gk2 > gk3 > gk4). Gk3_2_1 will try gk3 first, then gk2, then gk1. All the setting above can be added by port number, for examples: 168.56.9.22:8495 will call to 8495 port. srn, rsn: server. PSTN: route this call via PSTN line interface. This is usually used for the backup route for the rule setting on /Routing setup /VoIP Call out/ with “v” prefix.
user press the number is 886212345678 and the STRIP field is setting to 4, the first 4 digits 8862 will be truncated and actually call out number will be 12345678. g. Prefix: The numbers will be added on the prefix of the user press number. For examples, if user press the number is 12345678 and the PREFIX field is setting to 0028862, the actually call out number will be 002886212345678.
C.Setting Examples Here is some VoIP call out routing table setting examples below: a.
Port Message Display Port Type Talking Status Connected IP Caller ID Start Time End Time Dialed number name 1 FXS 2 Idle FXS 2004/02/19 2004/02/19 13:55:10 13:55:43 172.16.7.1 28 601 Strip Prefix (146)onHangup Idle c. Call by Domain name: Index Remark 2 Jack@SH Area Code Min Digits Max Digits IP Address VoipVoIP 8621 4 4 China 86 2 5 Profile Delete 01390244027 4 Gateway.dyndns.org 3 Release by Sec 33H33H 33H Delete 2 China.proxy.
Index Remark Area Code Min Digits Max Digits IP Address 1 Take_All 86 10.1.1.1 2 Never_Used 8621 20.1.1.1 Strip Prefix Profile Delete NEVER d. Strip and Prefix User is easy to combine using Strip and Prefix define to modify the phone number from phone to real call out phone number, for example, if the VoIP Gateway is installed on Taipei and use another Gatekeeper to service global service.
1 Via_GK2 1 gk2 2 gk2_3_1 3 gk 35H35H 35H Delete GK2_3_1 2 3 GK_ALL 36H36H 36H 37H37H 37H Delete Delete By Index 1, if user input any phone number with prefix code is 1, The VoIP Gateway will call out via Gatekeeper 2. By Index 2, if user input any phone number with prefix code is 2, The VoIP Gateway will try to call out by Gatekeeper 2 ( if register to Gatekeeper 2 is successful), if Gatekeeper 2 is not available, it will check Gatekeeper 3, then check Gatekeeper 1.
2 UsePF2 2 10.1.1.2 PF2 3 UseDefaultPF 3 Gate.proxy.com 41H41H 41H 42H42H 42H Delete Delete Q.931 Index Name VAD H.245 DTMF T.38 Package Tunneling Relay FAX Relay Frame CODEC Fast Start 1 2 ID1 AS ID2 AS ID3 AS ID4 AS PF1 ON G.723.1 ON Out band ON 3 ON 00001 H.323 1001 E.164 PF2 ON G.723.1 ON 00002 H.323 1002 E.
4.2.2.7 VoIP Call In Routing Table Setting /VoIP Setup/Routing Setup/VoIP Call In/ This page let you define the routing rule for Call in from VoIP. (VoIP Gateway got a VoIP call required form Network, and then translates the phone number passed from remote side VoIP Gateway to the real dial out number & line base on this VoIP Call In routing table).
b. Auth：Authorization IP check enable. Enable IP range authorization function. When Enable, the gateway will check the remote caller IP range setting on /VoIP Setup/Routing Setup/Authorization/, if it is within the authorization, the gateway will allow the call out, but if the remote caller’s IP is not in the range, it will refuse to call out and terminate this call. Area Auth Index CallWaiting Strip Prefix Maximum Minimum From To LineNo RS Verify Alert Profile Forward Delete Code 1 8862 c.
Index Area Code Strip 1 90 Prefix Maximum Minimum From To LineNo Gatekeeper Alert Profile Forward Delete 2 0,12345678 51H51H 51H Delete Ex: VoIP Call in number is 90 and real dial out number is 0,12345678 by stripe 2 digits and add 0,12345678 prefix, so the real dial out number is 0,12345678. e. Maximum: Define the maximum digits of call number allow to dial. If the length of dial number after pervious STRIP & PREFIX process is more than the setting, it will deny dialing out.
h. To: Define the ending line number for service this area code VoIP call. Index Area Code Strip Prefix Maximum Minimum From To LineNo Gatekeeper 1 601 1 1 2 602 2 2 Alert Profile Forward Delete 54H54H 54H 55H55H 55H 3 56H56H 56H Delete Delete Delete Ex. Any VoIP Call in number with prefix 601 will ring the line 1, and Any VoIP Call in number with prefix Call in number 602 will ring the line 2. any other numbers will ring any available (not busy) lines. i.
k. Call Waiting：Enable or Disable the call waiting function。 Enable: During Talk, you can answer another phone call and hold the original conversation. When some one call in when you are busy on another phone call , you will hear a du-du call waiting signal, please use flash key on your phone to hold the original call and answer the incoming call, press again flask key will switch back the original call party. Disable: Disable the Call waiting function.
ID1 AS ID2 AS ID3 AS ID4 AS PF1 ON G.723.1 ON Out band ON 3 ON 00001 H.323 1001 E.164 PF2 ON G.723.1 ON 00002 H.323 1002 E.164 1 2 Delete 66H66H 66H In band ON 3 Delete OFF 67H67H 67H Delete When VoIP call in number with prefix 1 will use the Profile named PF1 ( H.323 ID1 = 0001, E.164 ID=1001, DTMF Relay=Out band, Q.931 Fast Start=ON) to answer the VoIP Call in. When VoIP call in number with prefix 2 will use the Profile named PF2 ( H.323 ID1 = 0002, E.
10.1.1.1/104, it mean it will forward this call to IP 10.1.1.1 and calling number will change to 104). When the line 1 user is going to have a tour to another location with same VoIP Gateway equipment, user setup the called prefix number 601 forward to profile name CF1, and in CF1 profile, the Always field is set to hk.big.com/301, that mean any call number with prefix 601 will always be forward to another VoIP Gateway locate at hk.big.com and dial out number is 301.
4.2.2.8 VoIP Call In IVR Setting /VoIP Setup/Routing Setup/VoIP Call In IVR/ When Enable the [Prompt Voice for VoIP Call In function on /VoIP Setup/Advance Setup/Prompt Voice/ , all the remote party of VoIP caller will hear the customization upload voice file and need press the destination number. All the input number will be checked the number length and be strip/add prefix defined on this page.
A. Time & Digits wait for user The VoIP Gateway wait user input the number digits & time parameters as below: Time & Digits wait for user Press. e. MaxDigits: Define the maximum digits wait for user press for all VoIP Call Out, if user press digits match the number defined here. It will go to translate for call out rule without needed to press # key. f. FirstDigitTime: Define the waiting time (seconds) for user press phone number first digit.
c. Min Digits: define the minimum digits wait for user press for number fit this rule, if user press digits less the number defined here. It will keep waiting for input until exceed the FirstDigitTime defined time. If user press digits more then Min Digits here, the VoIP Gateway will wait time defined on OtherDigitTime then go to translate for call out rule without needed to press # key. d.
4.2.2.9 VoIP Routing Profile Setting /VoIP Setup/Routing Setup/Routing Profile/ This page defines the optional special VoIP parameters when making/received a VoIP call. For define some special parameters for different VoIP equipment or authorize purpose, please add a profile at /VoIP Setup/Routing Setup/ Routing Profile/ and use the same name as the profile on the Call in Routing Table (/VoIP Setup/Routing Setup/VoIP Call In/) or Call out Routing table (/VoIP Setup/Routing Setup/VoIP Call Out/). a.
When select In band to transfer the DTMF during VoIP, the user pressed DTMF tone will be treat as general voice and been compressed then transmit to remote side to decompress play back, it maybe cause some problem on duplicate or missing DTMF receive. When select Out band to transfer the DTMF during VoIP, the user pressed DTMF tone will be decode by local VoIP Gateway then transmit as signal, after received on received remote VoIP Gateway, it will be regenerate by remote VoIP Gateway.
Please remember to press the modify button to take it effect. For store back to flash memory, please press Save Modification (/System Maintenance/Save Modification/). Here is VoIP Routing Profile setting examples below: Q.931 Index Name VAD H.245 DTMF T.38 Package Tunneling Relay FAX Relay Frame CODEC Fast Start 1 2 ID1 AS ID2 AS ID3 AS ID4 AS PF1 ON G.723.1 ON Out band ON 3 ON 00001 H.323 1001 E.164 PF2 ON G.723.1 ON In band ON 00002 H.323 1002 E.
4.2.2.10 VoIP Forwarding Profile Setting /VoIP Setup/Routing Setup/Forwarding/ This page defines the forwarding behavior include: get an unmatched prefix number for VoIP call in, Line busy No answer Please add a profile at /VoIP Setup/Routing Setup/Routing Profile/ and put the name of profile on the Call out Routing table (/VoIP Setup/Routing Setup/VoIP Call Out/). a. Other: Define the forward IP and forward phone number when there is no match rule setting on VoIP Call Out Routing table.
f. No Answer Sec. Defined the wait seconds for redirect forward to another IP(or URL). g. Delete: Delete this rule item on routing table. h. To add new rule item on routing table, please assign the item number you want to insert before, input AREA CODE then press ADD button to add it on the list. Then modify the necessary information on the routing table list. Please remember to press the modify button to take it effect.
mean any call number with prefix 601 will always be forward to another VoIP Gateway locate at hk.big.com and dial out number is 301. When the line 2 is busy and another VoIP with prefix 602 Call in, it will forward it to the assist.big.com with same number (defined on OnBusy). When VoIP Call In number with prefix 602, it will ring the line 2 for 30 seconds (defined on No Answer Sec.), if no one answer line 2 within 30 seconds, it will forward the call to another VoIP Gateway located at assist.big.
1.4.2.11 VoIP Authorization Setting /VoIP Setup/Routing Setup/Authorization/ When this gateway has been used for outbound call, it can enable to check the remote caller gateway’s IP to decide accept or refuse the call. If define the IP range here and enable the [Auth] option on the/VoIP Setup/Routing Setup/VoIP Call In/ , only the IP in range will allow to call out by this gateway.
4.2.3 Register Server Setting 4.2.3.1 Register Status /VoIP Setup/Register Server/Register Status You Can check the register status of this gateway on this page. a. MAC: this gateway’ MAC Address。 b. RS1-4: Indicate the status of 4 server register. SIP&H323: The protocol used for registering the server, this gateway supports both H.323 and SIP protocol. Green Indicator: Successful to register server and the register phone number.
4.2.3.2 Setup the Register Server—SIP Protocol /VoIP Setup/Register Server/Server If you need use this gateway to register to the H.323 gatekeeper or SIP register/proxy server, you can setup the account for register here. This gateway can register up to four Servers simultaneously. a. Protocol: Select use SIP or H.323 protocol to register to server, by different protocol, the gateway will adjust the page for different parameters for input. b.
Disable Register SIP Proxy server function. d. SIP Proxy URL: Please input the IP/URL of the SIP proxy server. e. Port [1~65535]: Port number used for register to server. The SIP protocol default is 5060, please make sure you have same port number setting on the gateway and server. f. Thought Outbound Proxy: When your gateway is installed behind the firewall or NAT, you maybe need use Proxy server to relay your call. If so, please input the Outbound proxy server’s IP here. g.
q. Conference ID: Some SIP Server requires an ID to enable the conference function, please input the ID here to enable that. r. Enable: Enable or disable independently each line for register.
4.2.3.4 Setup the Register Server—H.323 Protocol /VoIP Setup/Register Server/Server When Select use H.323 to register gatekeeper, please input the flow information for register: s. Register Method: Global: All the lines of the gateway share same account to register. Independent: Each line independently for register. can set different/same account a. Enable H323 Gatekeeper : Enable Register H.323 Gatekeeper function. Disable Register H.323 Gatekeeper function. b.
g. FXO: Analog phone interface for connect to PSTN or PBX extension line FXS: Analog phone interface for connect to phone set or PBX Co. line. Remark: Remark for this routing rule. Please use UNDERLINE to replace the SPACE due to HTTP protocol limitation. h. E.164: phone number used for register to server. i. H.323: ID: Account name used for register to gatekeeper. j. Password: Password used for register to gatekeeper. k. Enable: Enable or disable independently each line for register.
4.2.4 Auto Provision function To use the auto provision function, the system have to install a dedicate Auto Provision Server for keep all parameters for installed gateways. When Enable the Auto Provision function, the System administer can modify all the Parameters of each gateway on the local Provision Server, and remote gateway will automatic download all the parameters from Provision Server. The Gateways can link up to five provision servers simultaneously for Redundancy backup the system.
It will indicate the status of linking to each Provision Servers. The gateway will link to one of the five setting Provision Server each time. If successful link, it will display green indicator. If failure link, it will display red indicator. The yellow indicator means it is disable. c. Last link time: Display the Date & Time for last successful link to Provision server on this gateway. d. Numbers of successful link：Display the times of successful linking from Provision Server to this gateway. e.
4.2.5 Advance Setup 4.2.5.1 NAT Traversal /VoIP Setup/Advance Setup/NAT Traversal If your VoIP gateway is installed behind NAT, you may need a special configuration and server to establish the VoIP communication, this gateway support several method for NAT Traversal as below: By Outbound Proxy: User can appoint an Outbound Proxy Server to handle the NAT traversal on VoIP Setup/Register Sever/Server #/ Declare NAT IP address: Select to enable the input the NAT VoIP Gateway IP of the network.
4.2.5.2 Listen Port /VoIP Setup/Advance Setup/Listen Port In this page, user can define the usage port for setup the VoIP communication. Both side of gateways need use the same port for begin VoIP communication. a. SIP Listen Port : Define the listen port for SIP protocol, the default port is 5060, input range from 1024 to 65535. b. H.323 Call Signal Port: Define the Call signal port for H.323 protocol, the default port is 1720, input range from 1024 to 65535. c. .H.
4.2.5.3 VoIP Package /VoIP Setup/Advance Setup/VoIP Package User can define the parameters relative about VoIP package on this page. Jitter Buffer(ms): Define the Jitter buffer size, input range is from 20 to 200ms. VoIP DTMF Relay Mode: Define the relay mode for DTMF signal: In band: When local gateway detects a DTMF signal, it will not decode it . The DTMF signal will been compress/decompress as VoIP voice package.
RFC2833: Payload number for DTMF[96~127]: Define the DTMF token on RFC2833, input range form 96 to 127. RTP Packet NAT Detect By: User can select the packet transmit method when under the NAT network environment, by IP (NAT 255 fake IP), or transmit by TCP port, usually suggest select the option “By IP or By Port”. Extension Number SIP Header To: Use the number in the SIP header column as the dial out number.
4.2.5.4 RTP Packet Summary /VoIP Setup/Advance Setup/RTP Packet Summary On this page, user will know the RTP package summary about last VoIP call. Line#: number of line Using CODEC: ex.: G.723.1, G.729a Source IP: Remote side IP Source Port: Remote side port Packet Interval: interval time between 2 packets.(ms) Packet Send: number of packets sent. Packet Received: number of packets received. Packet Lost: number of lost packets.
4.2.5.5 Flash & Call waiting /VoIP Setup/Advance Setup/Flash & Call waiting On this page, user can define the parameters relative to the FLASH key and Call Waiting function. These functions usually work with PBX Token for flash key on VoIP(!): Define the token for flash key during VoIP protocol ( use “! “ by default). Flash Signal generate length : Define the pause time (ms) for one “,” symbol at /Routing Setting/VoIP Call Out/.
4.2.5.6 Gain /VoIP Setup/Advance Setup/Gain This page defines different function gain on the gateway. Gain when Dial tone phase： When phone off hook, user will hear the dial tone generated from the gateway, sser can adjust the play/record gain during this phase for stable DTMF detection. After connection, the gains setting here is no use, the gateway will adjust the gain setting on /Line Configure/Line setting/ Play: Transmit gain from network to line. Adjust the speaker volume on the handset.
This item allows user to adjust the talking volume of the device, which means it will affects all ports of this device, User can select the db value of Play and Record, it will increase or decrease the talking and listen volume.
4.2.5.7 QoS /VoIP Setup/Advance Setup/QoS User can define the ToS field on the VoIP packet for Quality of Service control. The ToS field is included these 2 parameters: Precedence: bit 0,1,2 DSCP(Diffserv Code Point): bit 3~7 User can select input either IP Precedence or DSCP value. Or input the ToS binary code directly.
4.2.5.8 CDR /VoIP Setup/Advance Setup/CDR The Gateway can export all the CDR (Call detail Record) to external CDR server by HTTP protocol. The gateway supports up to 2 CDR servers for keep the record. Export to CDR Server: Please install and enable this function if you want to keep CDR of this gateway. CDR Server IP1: Please input the IP of first CDR server, if installed. CDR Server IP2: Please input the IP of second CDR server, if installed.
4.2.5.9 FoIP /VoIP Setup/Advance Setup/FoIP User can define the parameters relative FAX Over IP function. FAX Mode Appoint the FAX mode during FoIP: Disable: Only the VoIP function is supported on the gateway. FoIP is disabled. By default was disable. T.38: Enable and supported to T.38 protocol for FoIP . T.30: Enable and supported to T.30 protocol for FoIP, when to send or recived FAX information will through and under the G.711 codec . T.38+T.30: Enable and supported the T.38 and T.
Enable or Disable to send the double packet function during high speed T.38 FoIP.
4.2.5.10 Prompt Voice&Beep /VoIP Setup/Advance Setup/Prompt Voice & Beep This gateway can use voice or beep to prompt the user different situation. User can download/upload their own prompt voice wave files also.
Prompt voice for replace dial tone: Use a customize voice file to replace the dial tone Warming Prompt after VoIP out failure: Annunciate a customize voice file when VoIP call failure. No PSTN Line Connected : Annunciate a customize voice file when there is no trunk connected to the PSTN port and failure to call via PSTN line. Can not register to server warming prompt : Annunciate a customize voice file when the gateway failure to register to Register Server.
4.2.6 Application 4.2.6.1 Ping test /VoIP Setup/Application/Ping Test User can use the Ping Test function to test the network status or remote device Ping Destination:Ping: Target device IP for ping test. Number of Ping[1-100]Ping: Number of ping test, maximum is 100.
4.2.6.2 Telnet & SNMP /VoIP Setup/Application/Telnet & SNMP User can enable and set the user account for the SNMP and Telnet function of this gateway on this page. a. Enable telnet server Enable: Enable telnet service function, user can telnet this gateway. Disable: Disable telnet service function. b. Enable SNMP Server Enable: Enable SNMP service function, user can use SNMP on the gateway. Disable: Disable SNMP service function. c. d. e. User Name：Set a user name for Telnet & SNM login.
4.2.6.3 Call Timer /VoIP Setup/Application/Call Timer The gateway can enable to set the limit communication time to Line port by Area Code rule issue on this page. a. Pre Alarm: Set the Call time limit before the arrival, to issued the warning sound of few seconds down, default is disable, you can choose for four options(Disable, 10Sec, 20Sec, 30Sec). Line: Set enable or disable for the Line port . Remark: Remark for this routing rule.
4.2.7 4.2.7.1 System System Status This page reveals the status of the gateway including WAN, LAN and some hardware information. Internet This sub-block shows the Internet information of your home gateway. It depends on the WAN mode connecting to your ISP. The different items correspond to each WAN mode will be revealed after the common part of the Internet status sub-block. Common Part: Refresh Clicking this button, the browser refreshes the Internet status page to get the most update information.
Cable/DSL This field indicates the Internet connection status. Its value is Connected , Disconnected or Connecting. WAN IP Connected to the Internet through Cable or ADSL modem, the ISP will offer the home gateway a WAN IP address to communicate with other hosts in the Internet. Subnet Mask This field indicates a mask used to determine what subnet the WAN IP address belongs to. An IP address has two components, the network address and the host address. For example, consider the IP address 192.168.168.
Connection Type There are five ways to get the WAN IP address. They are DHCP, STATIC, PPPoE, PPTP and L2TP. This field indicates the way to get the WAN IP address. Through Figure 3-2 to 3-6 detail all specific items of each mode. Gateway IP Address This field is the LAN IP address of the home gateway. Subnet Mask This field is the subnet mask of the network in the LAN side. DHCP Server The home gateway supports DHCP service. DHCP Server.
Short for Media Access Control address, a hardware address that uniquely identifies each node of a network. This field indicates gateway’s LAN MAC address. 96H96H 96H 97H97H 97H WAN MAC Address This field indicates gateway’s WAN MAC address. Hardware Version Tells the version of hardware of the gateway. Serial Number This field indicates serial number.
4.2.7.2 System Settings This page is used to configure the names given by the ISP, if any, to represent the gateway, and also to set the local time zone. Host Name Some ISPs request the host name to represent the home gateway. Fill the host name given by the ISP, or you may not be able to access the Internet successfully. The maximum length of the host name is 32 bytes. Domain Name User configured domain name of the network maintained by gateway.
4.2.7.3 Date & Time This page is used to setting the system time of VoIP gateway; it can define the correct time by which ways.
4.2.7.4 Administrator Setting This page allows you to change the user and administration password used to manage this VoIP Gateway for security reasons.
4.2.7.5 System Log The system log page shows the gateway’s activity logs such as the Internet connectivity, hacker attack, intrusion detection and the wireless association. The log helps you to do fault analysis or regular statistics. Download Save the log to a local file. Clear Clear the log. Refresh Retrieve the log from the gateway and show in the text area again. System log The text area shows system activities. Security log The text area shows hacker attack or firewall logs.
Remote Log Settings Remote Log Check this file and the gateway will send the log message to the “Send Log to” remote host. Send Log to Set the IP address to send log. Email Log When the log buffer is full, the gateway will check whether this item is enabled. If enabled, the gateway will send all the log messages to the “Send Email to” email address. SMTP Server Set the SMTP Server (email server) to send the email log to. server’s name or its IP address.
4.2.8 Route Function(/System Setup) 4.2.8.1 Setting WAN connection: System Setup/Connection /System Setup/WAN/Connected Type Connection Type There are five ways to connect to the Internet. They are Dynamic IP, Static IP, PPPoE, PPTP and L2TP. The cable modem ISP usually requests you to obtain the WAN IP dynamically. Some ISPs request you to fill the host name. To do this, please go to “System/System Settings” page to change the host name. A.
Request IP address You can specify the IP address you desired. But the ISP has the right to neglect it and provides you a different one. MTU (576-1500) You can specify the MTU (maximum transmission unit) of your home gateway. The default value is 1500 bytes and in normal case, you don’t have to change. MAC Cloning Some ISPs will identify the MAC address registered by the user. If not registered, the ISP won’t allow the traffic to pass.
IP address assigned by your ISP Set the IP address that assigned by the ISP. Subnet Mask Set the subnet mask of the network. ISP Gateway Address Set the ISP’s gateway IP address. This address routes packets to Internet. C.
This page is the PPPoE configuration page. Most of the ISPs request the user to connect to central office (CO) side via PPPoE, acronym of Point-to-Point Protocol over Ethernet, which provides authentication, authorization and accounting. User Name Enter the user name provided by your ISP to identify the computer using PPPoE. Password Enter the password provided by your ISP to identify the computer using PPPoE. Please retype your password Retype the password to make sure type correct password.
that the user does not have to worry about the MTU size, the gateway routing engine will handle the MTU differences between PPP and the LAN Ethernet side. But for some old PPPoE server, you have to make the MTU size of the PPPoE side smaller than the default value, or some Web side is not able to access. PPPoE MTU should be set between 546 and 1492. Maximum Idle Time (60-3600) Set a period of time to disconnect PPPoE connection, when user’s idle time greater than it.
User can set this device work as a bridge, set the necessary information. A network bridge connects multiple network segments at the data link layer (layer 2) of the OSI model, and the term layer 2 switch is often used interchangeably with bridge. Bridging group IP address Enter the bridging IP address in this column. Bridging group Subnet Mask Enter the bridging Subnet Mask in this column. Bridging group Gateway Address Enter the bridging gateway IP address in this column. E.
Static DNS Server Check this item to make the primary and secondary DNS server at the next two rows active. If this field is not enabled, the statically configured DNS server will not take effect. Domain Name Server (DNS) Address Set your primary DNS in this field. Secondary DNS Address (optional) Set your secondary DNS to use when the primary DNS does not work.
4.2.8.2 LAN Settings /S/System Setup/LAN/LAN Settings/ The home gateway is an IP sharing device, which provides the home users share the same public IP address. While in the LAN side, each network device must have one private IP address to do network communication. This page is to set the configuration of the LAN interface of the gateway. IP Address Set this to be gateway’s LAN interface IP address.
IP Pool Ending Address The ending address provided by DHCP service. Lease Time Set the lease time of the IP address to renew. Local Domain Name Set the gateway’s local Domain Name. DNS Proxy The home gateway acts as a DNS Proxy. In this case, the DHCP service will set LAN interface IP address as the DNS server address, and inform the clients in the DHCP renew process. DHCP Client List This page lists all the DHCP clients in the LAN side. The DHCP server is capable of administering 253 clients.
This column shows the host name of the DHCP clients. IP Address This column shows the IP Address of the DHCP clients. MAC Address This column shows the MAC Address of the DHCP clients. Static Check this item to make the IP address static, and every time the client connect to the home gateway, it will get the same IP address all the way. Refresh Click this button, the browser request the DHCP client list from the home gateway and refresh the page again.
4.2.8.3 NAT Virtual Server This is the virtual server page, which set the rules translate the private IP/private port pairs to the public ports. Private IP Set the IP address of the virtual server. Private Port Set the port of the virtual server to connect to Internet. Type Set the protocol of the virtual server. Valid options are TCP, UDP and Both. Public Port Set gateway’s WAN port to connect to virtual server. Comment Let you put some notes to describe this entry. Enabled Enable this entry.
Special Application Trigger Port To define the TCP/UDP port range to monitor, when initiating from the home network, the public ports configured in this entry will be activated. Trigger Type Defines the protocol type: TCP, UDP or both of the trigger ports. Public Port This column defines the public ports opened when the trigger ports were seen. You can enter multiple ports, delimited by comma, or a range of port by dash. White space will be neglect.
The Port Mapping page does a port range mapping. Server IP Set one of LAN client’s IP address to do port mapping. Mapping Ports Set the ports to mapping from WAN to LAN. You can enter multiple ports, delimited by comma, or a range of port by dash. White space will be neglect. Public Type Defines the protocol type, TCP/UDP/Both, of the public ports. Comment Let you put some notes to describe this entry. Enabled Enable this entry.
ALG Some applications have to do Application Level Gateway (ALG) to monitor the transaction or monitor the payload of the packet. This page let you setup them. We use mnemonic nouns to describe the specific ALGs. The checked entries will be enabled by the ALG processing. For the Non-Standard FTP Port, you have to set the port number to let the ALG take it as an FTP session.
DMZ Check this item to enable DMZ service. DMZ table This table lets you configure the DMZ hosts. If you have more than one public IP Addresses, you can specify each one with an associated DMZ host. Public IP MZ table Address Select among the WAN IP addresses of the home gateway. IP Address of Virtual DMZ Host Fill the client in the LAN side as the associated DMZ host. Add Add this entry to virtual DMZ host table. Delete Delete this entry from virtual DMZ host table.
4.2.8.4 Firewall Firewall Options Enable Hacker Attack Protect The hackers always try to break into you system or crack down your network. The gateway provides hacker attack protect modules to protect your home network. Discard PING from WAN side “Ping” is the most widely used tool to diagnose the network. Hackers use “ping” to discover your gateway firstly and try to hack your home network later. Discard “ping” from WAN side prevent the hacker from discovering your home network by using “ping”.
scans all the TCP/UDP port of a station to find out the opened ports. After confirmed, the hackers try to connect to the listened ports to attack your computer. Check this option, the home gateway will drop the port scan packets to protect your system. Allow to Scan Security Port (113) Some Linux email servers will try to detect the security TCP/UDP port (113). If you drop it, the Linux email servers will not allow you to log into. Allow it can solve such kind of problem.
Client Filtering Enable Client Filter Check this box and all the entries below will take effect. IP Block the IP addresses ranged to connect to Internet. Port Set the port range, within which is not allowed to connect to Internet. Type Set which protocol, TCP/IP/Both, of this entry to inhibit. Block Time Two options are provided for this column, the “Always” and “Block”. If “Always” is selected, the next two items will be neglect.
URL Filtering Enable URL Filter Check this box and all the entries below will take effect. IP The IP addresses ranged will be the candidate to check the URL when they use web browser to access the Internet. URL filter string The specific URL string to block. Enable Check this item to enable this URL filter entry.
MAC Control MAC Control MAC Address Control Check this item to enable MAC control service. Filter out or only accept the following MAC address connect to Internet Select “Filter out”, the MAC addresses list in the following table will be block out to access the Internet. Or “Accept” to allow them to connect to the Internet freely. Configure MAC Address MAC Address List all the MAC address want to block then connect to Internet. Comment Put your note why you set here.
4.2.8.5 Routing Routing Table This field indicates the destination IP of this route entry. Subnet Mask This field indicates the Subnet Mask of this route. Gateway If a packet’s destination IP address do “bit and” operation with the “Subnet Mask” equals to the “Destination LAN IP”, the packet will send to the Gateway. Interface This field indicates which network interface deal the connection. Refresh Refresh the routing information from the home gateway again.
Static Routing Destination LAN IP This field indicates the destination IP of this route entry. Subnet Mask This field indicates the Subnet Mask of this route. Gateway If a packet’s destination IP address do “bit and” operation with the “Subnet Mask” equals to the “Destination LAN IP”, the packet will send to the Gateway. Add Click this button to add this entry to routing table statically. Delete Click this button to delete this static entry.
4.2.8.6 UPnP Settings UPnP Enable UPnP Check this item to enable UPnP. UPnP Port Number Set UPnP port number to announce to the UPnP control points. The UPnP control points use this TCP port to send request to the home gateway. Advertise Time (60-1800) Set the time interval after which the home gateway sends advertisement packets. Subscribe Timeout (60-1800) The UPnP control point subscribes a request to the home gateway.
Port Mapping Remote Host This field lists remote host that connect to LAN client. External Port This field lists the port of the remote host connect to LAN client. Internal Client This field lists LAN client connect to the Internet. Internal Port This field lists port of LAN client connect to the Internet. Protocol This field lists the protocol, TCP/UDP, of the connection. Duration This field lists the duration of the connection. Description This field lists the description of list port mapping.
4.2.8.7 DDNS Enabled Check this item to enable the DDNS settings Disabled Check this item to disable the DDNS settings Host Name Set your host name that need to do DDNS update. DDNS Server Set the DDNS server that updates your IP address. User Name The DDNS server requires you to supply a name and password to update your IP address. You have to register the user name and password offline to the specific DDNS server.
4.2.9 Bandwidth & VLAN 4.2.9.1 Bandwidth Control You can enable WAN or LAN bandwidth control in this configuring page, so far only the Upsteam Bandwidth is workable, you can fill in the limit value in Upstream Bandwith to make limiytation. 4.2.9.2 VLAN This function is used for special purpose of application. For more details about VLAN, please refer to application note for VLAN, which is in the end of this manual.
4.2.10 Backup/Restore 4.2.10.1 Configuration /Backup-Restore/Configurations/ This page let you backup / Restore all of your configuration parameters on the VoIP Gateway. It is very good idea to back up all of your VoIP Gateway configuration parameters after install. a. To Backup, press Download setting backup file, and input the file name you want and file location to save. b. To Restore, press the Browse button the select the backup configuration parameters file to upload then press Restore .
4.2.10.2 VoIP Module /System Maintenance/Backup-Restore/Configurations/ This page displays the current firmware module version and let you backup / Restore all of your VoIP firmware module on the VoIP Gateway. Please use this page to update the VoIP module firmware if necessary. a. To Restore from local file, press the Browse button the select the VoIP module file to upload then press Restore .
4.2.10.3 Reboot System /Reboot System Use the Reboot button on this page to reboot your VoIP Gateway, before you reboot, please make sure you have to press the Saved modification to save your current configuration to Flash ROM, otherwise all the change will be disappear after reboot.
4.2.11 Save Modification /Save Modification/ Most of the VoIP Gateway parameters will take effective after you modify, but it is just temporary stored on RAM only, it will disappear after your reboot or power off the VoIP Gateway, to save the parameters into Flash ROM and let it take effective forever, please remember to press the Save Modification button after you modify the parameters.
