User’s Manual – Version: 2.
Table of Contents TABLE OF CONTENTS 2 PREFACES 5 0.1 ABOUT THIS MANUAL 5 0.2 COPYRIGHT DECLARATIONS 5 0.3 TRADEMARKS 5 0.4 SAFETY INSTRUCTIONS 5 0.5 WARRANTY 5 INTRODUCE 6 1.1 OVERVIEW 6 1.2 ACRONYMS TABLE 6 1.3 INTRODUCTION 7 1.4 FRONT PANEL LED INDICATORS & REAR PANELS 1.4.1 VOIP GATEWAY & SIP PROXY SERVER & GATEKEEPER OUTLOOK 1.4.2 FRONT PANEL LED AND CONTAINER DESCRIPTIONS 1.4.3 REAR PANEL DESCRIPTIONS 8 8 9 12 1.5 FEATURES AND SPECIFICATIONS 1.5.1 VOIP GATEWAY FEATURES 1.5.2 H.
WIZARD FOR QUICK SETUP 32 3.1 WAN PORT TYPE SETUP 32 3.2 CONFIGURING NAT OR BRIDGE SETTING: 34 3.3 VOIP CALL PROTOCOL SETUP 35 GATEWAY SETTING 36 4.1 NETWORK CONFIGURATION 4.1.1 WAN PORT T YPE SETUP 4.1.2 CONFIGURING LAN IP ADDRESS AND DHCP SERVER 4.1.3 VIRTUAL SERVER SETUP 4.1.4 DYNAMIC DNS 4.1.5 NETWORK MANAGEMENT 38 38 40 41 41 42 4.2 VOIP SETUP 4.2.1 H.323 SETUP 4.2.2 SIP SETUP 4.2.3 DIRECT CALL (PEER TO PEER) SETUP 4.2.4 OTHER VOIP SETTING 43 43 51 57 60 4.
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PREFACES 0.1 About This Manual This manual is designed to assist users in using VoIP Gateway and Call Manager. Information in this document has been carefully checked for accuracy; however, no guarantee is given as to the correctness of the contents. The information contained in this document is subject to change without notice. 0.2 Copyright Declarations Copyright 2006 Telephony Corporation. All rights reserved. This publication contains information that is protected by copyright.
warranty. We reserve the right to revise the manual and online documentation and to make changes from time to time in the contents hereof without obligation to notify any person of such revision or changes. Note Repair or replacement, as provided under this warranty, is the exclusive remedy of the purchaser. This warranty is in lieu of all other warranties, express or implied, including any implied warranty of merchantability or fitness for a particular use or purpose.
control protocol) SIP STUN TCP UPnP Session Initiation Protocol Simple Traversal of UDP through NATs Transmission Control Protocol Universal Plug and Play 1.3 Introduction SLIC URI Subscriber Line Interface Circuit Uniform Resource Identifier UDP VoIP User Datagram Protocol Voice Over Internet Protocol This VoIP Gateway provides a total solution for integrating voice-data network and PSTN. The L200 and S series gateway is low to high density port gateway which support SIP / H.323 VoIP Protocol.
S200 Series ( 2 analog lines gateways/ Embedded H.323 Gatekeepers/ Embedded Sip Proxy Servers) S200 0 2 4 1 √ √ SK200/SVR200 S201 1 1 4 1 √ √ SK201/SVR201 S202 2 0 4 1 √ √ SK202/SVR202 L200 Series ( 2 analog lines Gateways/embedded Gatekeepers/Sip Proxy Servers) L200 0 2 1 1 √ √ LK200/LVR200 L201 1 1 1 1 √ √ LK201/LVR201 L202(*) 2 0 1 1 √ √ LK202/LVR202(*) 4 0 4 1 √ √ Call Manager C400 * : manufacture by order (lead time : 60 days) 1.
L200 Series: -L200 Series Gateway -LK200 Series Gateway building in H.323 Gatekeeper Software -LVR200 Series Gateway building in SIP Proxy Server Software 1.4.2 Front Panel LED and Container Descriptions L200(GW/GK/SVR) Series -----------------------------------------------------------------------------------LED State Description -----------------------------------------------------------------------------------1.
-----------------------------------------------------------------------------------LED State Description -----------------------------------------------------------------------------------1. POWER On GW is power ON Off GW is power Off ------------------------------------------------------------------------------------2.
ACT On LAN is connected successfully Flashing Data is transmitting Off Ethernet not connected to PC ------------------------------------------------------------------------------------5. FXS(Port) Off Telephone Set is On-Hook Flashing Ring Indication On Telephone Set is Off-Hook ------------------------------------------------------------------------------------6. FXO(Port) Off Line is not enabled On Line is busy ------------------------------------------------------------------------------------7.
Flashing Ring Indication On Telephone Set is Off-Hook ------------------------------------------------------------------------------------6. FXO(Port) Off Line is not enabled On Line is busy ------------------------------------------------------------------------------------6. LCD Panel Off System is Shutdown On System is Up NOTE: System initialization will turn some LEDs ON for a few sec. When System Boot/Reboot , the Port LEDs will flash in turn for a few sec. 1.4.
S200/S400(GW/GK/SVR) Series & C400 Call Manager Item 1 Port FXS(Foreign Exchange Station) Description Connect to Phone with RJ-11 (Black) analog line. FXS port was connected to your telephone sets, FAX, or Trunk Line of PBX. 2 FXO(Foreign Exchange Office) Connect to PBX or CO line with RJ-11(Write) analog line. FXO port was connected to the extension port of a PBX or directly connected to a PSTN line of carrier. 3 WAN(Wide Area Network) Connect to the network with an Ethernet cable.
S800 (GW/GK/SVR) Series & C800 Call Manager Item 1 Port FXS(Foreign Exchange Station) Description Connect to Phone with RJ-11 (Black) analog line. FXS port was connected to your telephone sets, FAX, or Trunk Line of PBX. Connect to PBX or CO line with RJ-11(Write) analog line. FXO port was connected to the extension port of a PBX or directly connected to a PSTN line of carrier. Connect to the network with an Ethernet cable. This port allows your GW to be connected to an Internet Access device, e.g.
SB800/S1600/S2400 Series Gateway Item 1 Port Standard Telco 50 PIN Connector (RJ-21) Description It is a 50 pins RJ-21 connector for connecting to telephone patch pane 2 RES(Reset button) The reset button, when pressed, resets the cable voice gateway without the need to unplug the power cord. Push this button until 5 seconds, and GW will be set to factory default. 3 RS-232 RS-232 console port connect to PC, Use Pc com port to connect RS-232 console , setting GW configure.
1.5 Features and Specifications The L200 and S Series Gateways provide many built-in server and software features to provide a convenient comprehensive solution for your VoIP network 1.5.1 Gateway Features VoIP Key Features Both support SIP and H.323 protocols: SIP Registration and Digest Authentication and H.323 Gatekeeper Registration. Single Number / Account for multiple ports. Caller ID Delivery and Detection: FXS support DTMF&FSK Caller ID generation; FXO supports DTMF&FSK Caller ID detection.
1.5.2 Embedded H.323 Gatekeeper Features 200 H.3232 endpoints scale: SK400 providing 250 H.323 endpoints to register. Register Security Policy: SK400 providing security setting on your H.323 VoIP network. This provides protection for VoIP calls and insures proper endpoint identification. Pre-Granted Endpoints: letting other gateways or H.323 endpoints which were not register to this embedded gatekeeper. And the registered VoIP Gateways can make an Off-Net call to these pre-granted endpoints..
STUN (RFC3489), ENUM (RFC 2916), RTP Payload for DTMF Digits (RFC2833), Outbound Proxy Support. Support Virtual Server, DHCP Server. LAN : WAN: Network Address Translation: Support PPPoE client, DHCP client, Fix IP Address, DDNS client. Providing build-in NAT router function. Smart QoS: Guarantee the voice bandwidth TOS: IP TOS (IP Precedence) / DiffServ General Specification AC power : AC100V-240V, DC12V/1.
Installation and Setup 2.1 Package Content Please check enclosed product and its accessories before installation. (Refer to the item number). These contents are from pre-released product. The contents for the final product might change a little bit. Appurtenances: Item Appurtenances Description 1 CD-ROM CD Include in all product user manual and datasheet. 2 RJ-45 cable Internet cable RJ-45 connect to NIC/Gateway/Router 3 RS-232 cable RS - 232 Console port connect to PC COM port.
2.1.1 L200 /S200/S400 Series Gateway & embedded Gatekeeper/ Sip Proxy Server The 2/4 Port packet contents: GW,GK,SVR(S200/ S400 Series) RJ-45 AC Power Adapter CD-Rom(User manual) 2.1.
2.1.3 SB800/S1600/S2400 Series High Density Gateway Patch Panel: The 8/16/24 Port packet contents: Gateway (SB800/S1600/S2400 Series) RJ-45 RS-232 AC Power Cable CD-Rom(User manual) Patch Panel(Option) X1 X1 X1 X1 X1 2.1.
Call Manager contents: Call Manager(C400/C800) RJ-45 RS-232(only C800) AC Power Adapter CD-Rom(User manual) Modular Duplex Jack Extension Cord X1 X1 (X1) X1 X1 X4 (C400) X8 (C800) 2.2 Installation Install Gateway 1 Connect the 12V DC IN to the power outlet with power adaptor. 2 Connect FXO to PSTN / PBX Extension Line. 3 Connect FXS to a telephone jack with the RJ-11 analog cable (Phone / PBX Trunk Line.) Connecting to a PC: 1 Connect the Ethernet cable (with RJ-45 connector) to any LAN port.
LAN(Local Area Network) RES(Reset button) AC power(DC in 12V) allows your GW to be connected to an Internet Access device, e.g. router, cable modem, ADSL modem, through a networking cable with RJ-45 connectors used on 10BaseT and 100BaseTX networks. Connect to PC with Ethernet cable. 1 port allows your PC or Switch/Hub to be connected to the GW through a networking cable with RJ-45 connectors used on 10BaseT and 100BaseTX networks.
2.3 Setup There are 3 way to setting gateway - [Web User Interface] [Telnet] [Console] (Some Series modules have RS-232 console port like S800/SB800/S1600/S2400). 2.3.1 Factory Default setting WAN Port IP address : 192.168.1.1 LAN Port IP address : 222.222.222.1 LAN DHCP Server enable IP range: 222.222.222.51 ~ 222.222.222.
5. 6. 7. 8. Make connection(Bits Pre second:38400 Flow contact: None) Input “Enter” and Show Welcome display. Login, input the Password to login.(Password as the same as Access, default is admin) Setting Gateway Configure like telnet mode (Setting Table following as Telnet Setting table) 2.3.3 Telnet Connect WAN/LAN port to Internet or PC and gateway at the same subnet. you can use telnet remote to configure your gateway. 1. Connect Gateway online (Wan/Lan) 2. Remote Gateway by Telnet.
3. Input Password (Gateway Access password, Default: admin), If login successful, you will enter the welcome display. (1:Gateway Model 2:Firmware Version 3:Wan/Lan Status 4.DDNS Status 5:VoIP Status) 4. Gateway Telnet Setting Table, Use 1~9 a~z select setting, “ESC” is back setting. Item Setting Option Main [1]Advanced Setup [1]Advanced Setup ……1.WAN Setting [1]Advanced Setup ……2.LAN Setting [1] Advanced Setup. [2] System Administration. [3] Save Current Configurations. [4] Upgrade Software. [5] Ping.
[1]Advanced Setup ……3.Virtual Server [1]Advanced Setup ……4.Dynamic DNS [1]Advanced Setup ……5.Network Management [1]Advanced Setup ……6.VoIP Basic [1]Advanced Setup ……7.Dialing Plan [1]Advanced Setup ……8.VoIP Advance Setting 1.Add Virtual Server 2.Delete Virtual Server 1.Change DDNS username 2.Change DDNS password 3.Change DDNS domain name 4.Change DNS server IP 1.Change web server port 2.Change telnet server port 1.Change VoIP Protocol to H.323 2.Change Port Number/Account/Password 3.
[1]Advanced Setup ……9.Hot Line Setting [2] System Administration. (3)Network Advance 1.Disable Smart QOS 2.Bandwidth Control 3.G.723 Bandwidth 4.G.729 Bandwidth 5.Set IP TOS 1.Change Port1 Hot Line Number 2.Change Port2 Hot Line Number…….(To your own port) 1.Save Configuration 2.Access Control 3.Set to Default 4.System Information 5.NTP Setting 6.Syslog Setting 2.3.4 Web User Interface The gateway has a built-in HTTP(Web) server for configuration.
(Your particular system will be different from the screen shown here.) Check that you have an Ethernet network card installed. If not, refer to the card manufacturer’s documentation and install the card and drivers. If your card is installed, 1. Click the Add button. The Select Network Component Type dialog box will open. The box will show four options: Client, Adapter, Protocol, and Service. 2. Select Protocol and click the Add button. The Select Network Protocol dialog box will open. 3.
Configuring the TCP/IP Protocol 1. On the Network dialog box Configuration card, select TCP/IP and then click Properties.” The TCP/IP Properties dialog box will open. 2. On the IP Address tab, click Obtain an IP address automatically. As the DHCP (Dynamic Host Configuration Protocol) server built into the router is enabled by default, your computer will get an IP address, subnet mask, and other related IP network settings from the router. 3. On the DNS Configuration tab, click Disable DNS”. 4.
Connecting to the Web Configuration via a Web Browser 1. Launch the Web browser(IE or Firefox). Enter http://222.222.222.1 into the browser Address window and press the Enter Key 2. An authentication dialog box will open. 3. If this is a first time setup of the router, type “admin” as the User Name and the Password field as “admin”. Click OK.
4. The Web Configuration Setup Main Menu will open. On the main page [Setup Wizard], [Advanced Setup] and [System Information] were displayed. Wizard for Quick Setup Wizard for Quick Setup gateway, After finishing the authentication, the Main menu will display 3 parts of configuration, please click “Wizard Setup” to enter quick start: 3.1 WAN Port Type Setup For most users, Internet access is the primary application. The S Series Gateway support the WAN interface for Internet access and remote access.
Fixed IP User: If you are a leased line user with a fixed IP address, fill out the following items with the information provided by your ISP. IP Address: check with your ISP provider Netmask: check with your ISP provider Default Gateway: check with your ISP provider ADSL Dial-Up User (PPPoE Enable) Some ISPs provide DSL-based service and use PPPoE to establish communication link with end-users. If you are connected to the Internet through a DSL line, check with your ISP to see if they use PPPoE.
DHCP Client (Dynamic IP): Get WAN IP Address automatically IP Address: If you are connected to the Internet through a Cable modem line then a dynamic IP address will be assigned. 3.2 Configuring NAT or Bridge setting: Bridge Mode: Select S series Gateway as bridge. NAT mode: LAN IP Network Configuration IP Address: Private IP address for connecting to a local private network (Default: 222.222.222.51). Subnet Mask: Subnet mask for the local private network (Default: 255.255.255.0).
3.3 VoIP Call Protocol Setup Step1 : Configure VoIP Call Signal Protocols : User could select H.323 or SIP Protocol, and click “select” Step2 : configure the numbering with phone/line ports. Phone Number (FXS): The representation number is the phone number of the telephone that is connected to Phone port. Line Number (FXO): Line ports are connected to the extension ports of the PBX system or the PSTN line. They have a common Line Hunting Group Number.
“Leading Number” is the leading digits of the dialing number. “Min Length” and “Max Length” is the min/max allowed length you can dial. “Strip Length” is the number of digits that will be stripped from beginning of the dialed number. “Prefix Number” is the digits that will be added to the beginning of the dialed number. “Destination” is the IP address of the destination Gateway that owns this phone number. Step 5: Finishing the Wizard Setup After completing the Wizard Setup, please click “Finish” bottom.
Advanced Setting Label Network Setup Label WAN Setting LAN Setting Virtual Server Dynamic DNS Network Parameters Sets/changes the WAN port Type like “Fixed IP”, “DHCP Client” or ”PPPoE”. Modifies the IP address of the LAN port and setting DHCP Server parameters. Remote user can access server such as Web or FTP at you local site via public IP address can be automatically redirected to local servers configured with private IP address.
Management Label Save Configuration You can save configuration and restart the gateway with the default configuration or with the current running configuration. Access Control Users can Sets/changes the administrator password.. Set to Default You can restart the gateway with the default configuration. Backup/Restore Configuration User can backup the configuration file of Gateway to PC or Restore the configuration file from PC.
Static IP: You are a leased line user with a fixed IP address; fill out the following items with the information provided by your ISP. IP Address: check with your ISP provider Subnet mask: check with your ISP provider Default Gateway: check with your ISP provider PPPoE for ADSL Some ISPs provide DSL-based service and use PPPoE to establish communication link with end-users. If you are connected to the Internet through a DSL line, check with your ISP to see if they use PPPoE.
IP Address: If you are connected to the Internet through a Cable modem line then a dynamic IP address will be assigned. (Note : WAN port display the IP address, Subnet Mask and Default gateway IP address if DHCP client is successful) 4.1.2 Configuring LAN IP Address and DHCP Server There are two kinds of network feature to configure: Bridge Mode and NAT Mode Bridge Mode: Select this Gateway as Bridge. Let gateway Lan port like Switch/HUB.
users (PCs) on Yes: Enables the DHCP server. No: Disables the DHCP server. Start IP Address: Sets the start IP address of the IP address pool. End IP Address: Sets the end of IP address in the IP address pool. DNS Server IP Address: DNS stands for Domain Name System. Every Internet host. must have a unique IP address, also they may have a human friendly, easy to remember name such as www.yahoo.com. The DNS server converts the human friendly name into it’s equivalent IP address.
dynamic IP addresses, such as those assigned by an ISP or other DHCP server. DDNS is popular with home network, who typically receive dynamic, frequently-changing IP addresses from their service provider. To use DDNS, one simply signs up with a provider and installs network software on their host to monitor its IP address. How to use DDNS First: you should register a new DDNS service account from this web site: http://www.dyndns.
4.2 VoIP Setup Gateway support 2 VoIP protocol - H.323 / SIP, you can register to H.323 Gatekeeper or SIP proxy server. Gateway is not a softswitch, it only can use 1 VoIP protocol (SIP/H.323) at the same time! If you don’t register GK or Proxy server, you can make Peer to Peer call by IP address or domain name (Setting Dialing plan). 4.2.1 H.323 Setup Gateway H.323 protocol support H.323 (v2/v3/v4), H.225, Q.931, H.245 and RTP/RTCP. Don’t support H.235 security, can’t use H.
FXS Number: The representation number is the phone number of the telephone that is connected to FXS port. FXO Number: FXO ports are connected to the extension ports of the PBX system or the PSTN line. They have a common Line Hunting Group Number. When this number is dialed, the Gateway system will find a free FXO line connected to PBX. This hunting will skip all busy lines and absent lines and find only the idle line to the PBX. After the available line is found, you can hear the dial tone from PBX.
H.323 Parameters Label H.323 ID Primary Gatekeeper IP Address Secondary Gatekeeper IP Address Sets the unique name of this Gateway, that is communicated as part of H.323 messaging.. There are two gatekeeper address fields, one is primary, the other secondary. If this gateway does not want to register to any gatekeeper, just set value 0 to the primary gatekeeper address. If the primary gatekeeper address is not 0, the gateway will register to the primary gatekeeper.
H.323 VoIP Advanced Configuration There are many H.323, VoIP, Codec and other more detail Setting, you can set in “Advance Setting”. For SIP and H.323, there are a little different in advance setting. There are 3 different parts to setting about VoIP, Telephone and network. [Advance Setting] Item Description DTMF Relay for H.323: After the VoIP call is connected, when you dial a digit, this digit is sent to the other side by DTMF tone. There are two methods of sending the DTMF tone.
MAC Authentication: Watchdog: phone set. The H.323 call signaling part could be connected or alerting during this ringing period. If this selection is enabled, the H.323 signaling part is connected during the ringing period. The benefit of this situation is that the remote side could hear the status of the specific port. That is, the remote side will hear ring back tone if the Gateway is really ringing the phone set. If the phone set is busy, the remote side will hear busy tone.
Item Description Silence Compression: (VAD) If this function is enabled, when silence is occurred for a period of time, no data will be sent across the network during this period in order to save bandwidth. (If you use Asterisk, please disable Silence Compression, it maybe make you call disconnect.) The Codec is used to compress the voice signal into data packets. Each Codec has different bandwidth requirement. There are four kinds of Codec, G.723, G.729AB, G.711_u and G.711_A. The default value is G.
other Option. FXO Ringer Voltage Threshold: FXO Ringer Voltage filter: FXS Battery Reversal Generation: FXO Answer Supervision Line Silence Disconnect: FXO Answer Delay Time FXO Answer Mode Sometime you use FXO connect to PSTN/PBX can’t be pick up the call. PSTN/PBX can’t detection FXO voltage, you can adjust this function. Default is low.( Low : 16.5 Vrms Medium: 24 Vrms High: 49.5 Vrms) Some vendor’s PBX generates the leakage voltage from extension port. That will mislead the FXO become Off-hook status.
[Network Advance] Item Description Smart-QoS: If this function is enabled, when VoIP call is occurred, the other data will be automatically reduced traffic which across the internet in order to guarantee the voice bandwidth. Bandwidth control: You can configure your bandwidth what the Max byte of download and upload of ADSL modem rate. Setting G.723 / G.729 voice compression size. Quality and Packet size can adjust by you want. G.723/G.
4.2.2 SIP Setup Gateway SIP support SIP(RFC3261), SDP(RFC2327), RFC2833, STUN(RFC3489), Symmetric RTP, outbound proxy, ENUM(RFC2916),and RTP/RTCP.SIP NAT pass through Function can support 80% NAT/Firewall that you don’t setting DMZ/Virtual server in router or Firewall. 1. Select “SIP Protocol” 2. SIP number / account (username) and Password Setting: Please fill out the SIP account including username / password from ITSP.
SIP Hunting Table: This allows gateway can answer SIP call from internet by Hunting. For example: Port 2-7 are hunting for the Port 1 SIP account. When port 1 are on call, the other one SIP call from internet will ring port 2.S200/400/800 Series hunting table is use click box. 3. SIP Proxy Server setting, setting SIP proxy server register information. (If user does not need register SIP Proxy Server, Please go to Dialing Plan Policy) SIP Proxy Server Label SIP Proxy Server Setting 1.
4. If your gateway under the NAT/Firewall, you should setting different NAT Pass function. if you setting STUN/Outbound Proxy, you should have a STUN/Outbound proxy server. If they can’t pass NAT or one way talk happen, try to open “DMZ” and virtual server “5060” port in router. Symmetric RTP: default use Nat pass function. STUN Client: setting your STUN server information, default STUN server is FWD STUN server. Outbound Proxy Support: Setting your Outbound Proxy server information.
signal. RFC2833 Payload: FAX Mode Option: Watchdog: Adjust RFC2833 DTMF payload value, range from 96 to 127, default is 101. T.30/T.38 real-time FAX compliant Voice/FAX auto-switch. The T.38 is a “Real Time Group 3 Fax Communication over IP network” format. That’s meaning it’s a protocol for Fax over IP. You have to enable this function (T.38 mode isn’t support all gateway, different band use T.38 have a little change, it maybe let T.
Item Description Silence Compression: (VAD) If this function is enabled, when silence is occurred for a period of time, no data will be sent across the network during this period in order to save bandwidth. (If you use Asterisk, please disable Silence Compression, it maybe make you call disconnect.) The Codec is used to compress the voice signal into data packets. Each Codec has different bandwidth requirement. There are four kinds of Codec, G.723, G.729AB, G.711_u and G.711_A. The default value is G.
FXO Transmit Hybrid: FXO Ringer Voltage Threshold: FXO Ringer Voltage filter: FXS Battery Reversal Generation: FXO Answer Supervision: Line Silence Disconnect: FXO Answer Delay Time FXO Answer Mode ECHO cancellation adjust, 3 mode setting to solve ECHO problem, default is Mode 1. If you have echo problem, try to select other Option. Sometime you use FXO connect to PSTN/PBX can’t be pick up the call. PSTN/PBX can’t detection FXO voltage, you can adjust this function. Default is low.( Low : 16.
[Network Advance] Item Description Smart-QoS: If this function is enabled, when VoIP call is occurred, the other data will be automatically reduced traffic which across the internet in order to guarantee the voice bandwidth. You can configure your bandwidth what the Max byte of download and upload of ADSL modem rate. Setting G.723 / G.729 voice compression size. Quality and Packet size can adjust by you want.
In the “Outgoing Dial Plan Configurations” settings: Maximum Entries : 50 “Outbound number” is the leading digits of the call out dialing number. “Length of Number” has two text fields need filled: “Min Length” and “Max Length” is the min/max allowed length you can dial. “Delete Length” is the number of digits that will be stripped from beginning of the dialed number. “Add Digit Number” is the digits that will be added to the beginning of the dialed number.
“Inbound number” is the leading digits of the dialing number. “Length of Number“ has two text fields need filled: “Min Length” and “Max Length” is the min/max allowed length you can dial. “Delete Length” is the number of digits that will be stripped from beginning of the dialed number. “Add Digit Number” is the digits that will be added to the beginning of the dialed number. “Destination Tele port” is “Tel-port”; this is for local dial plan setting phone number.
Example3: Termination Call to FXO for One-Shoot Call Port 1: FXO was connected to PSTN (area code is 81xxxxxxxx). H.323 (SIP) leading number “081x”incoming, and delete the first one digit “0”, and call to PSTN number. Example4: Termination Call to FXO Port 1: FXS Port 1: FXO was connected to PSTN (area code is 92xxxxxxxx). Port 1 FXS call to “092x” to PSTN, FXO port will delete the first one digit “0” and call to PSTN number. ( [x]: mean wild card, 0~9.
Port Status: Each of port show status table. you can view all port status. like on/off hook , caller/callee IP, duration , and packet loss. (SB800/S1600/S2400 Series can change page to view all ports) Port Status Display : This selection will display concurrent call status of this Gateway. The status information of each voice channel includes codec, dialing number and destination IP address. The status is refreshed every 3 seconds.
4.3 SIP Proxy Server (SVR) Setup The SVR200/SVR400/SVR800 Series, a Hybrid SIP Proxy Server, registers and authenticates users, and routes calls between user agents. With SVR400/SVR200/SVR800 Series SIP Server, you can use SIP Agents, SIP soft phones, and SIP Gateways for VoIP communications. The SVR400/SVR200/SVR800 Series also provide Trunk Route capability to call Least Cost Route to ITSP or PSTN line.
2. Setting Proxy Parameter Register Expired Time(seconds): This field sets how long an entry remains registered with the SIP register server. The register server can use a different time period. The Gateway sends another registration request after half of this configured time period has expired. SIP Server Port: Setting the Proxy service Port, default port is 5060. 3. Setting Authentication, if you want to use MD5 Authentication , add user and password in table.
Enable/Disable MD5 Authentication: If you want to UA registered SVR have Authentication, Enable this option. Add Username/Password: Add Username / Password for Authentication. (Max 250 username / password and Max username/password is 30 digit) 4. If you want to view gateway registered information, select “Register UA” option. 5. If you want to view CDR information (Real Time), select “CDR (Real Time)” option. 6.
7. If you want to register Multi-SIP Proxy Server, select “trunk” option. ITSP Trunk: There are 4 trunk lines for user t configure. Each trunk have two types: A. ITSP registering: This SVR can register to another one ITSP Proxy Server for call international calls to save the call cost.
Termination Trunk: Setting the Proxy service Port, default port is 5060. This SVR can route the calls to terminate gateway for call termination. 8. If you want to set Dialing Rule for Call Routing, select “Outgoing Dial Rule” option. Outgoing Dial Rule Setting: SVR can register to another one ITSP Proxy Server for call international calls to save the call cost.
Least Cost Trunk Route: SVR200 / SVR 400 /SVR 800 series provide 4 Trunk Route Setting for Off-Net call to ITSP or Termination Gateway. (SVR series can subscribe to most 4 Service provider) For example, you can send all international calls through 4 different ITSP by the least call rate. The feature can select the most effective service provider by call route rule setting. 4.4 H.323 Gatekeeper (GK) Setup 2/4/8 port gateway can embedded H.323 Gatekeeper (GK). GK are both H.
3. Setting simple security, if you want to gateway authentication before registered, you can use this function. Gateway IP address: Input you allow registered gateway IP address, if gateway not in this IP address, they can’t registered. H.323 ID: Input H.323 ID for Authentication, gateway must send H.323 ID to GK check, H.323 ID just like “Password”. MAC address: Input you allow MAC address to authentication.
6. You can let GK to call other gateway without registered, setting in Pre-granted GW option. MAX setting 20 gateway. Gateway IP Address: Input you want to call without registered Gateway IP address Phone number: Input that gateway call phone number. 7. You can change your gatekeeper RAS port to gateway registered. 5 RAS port can be setting. RSA Port: Setting H.323 RSA signal port , you can setting 5 different RSA port to Registered.
4.5 Call Manager Call Manager is used to determine which mode your dial calls will go through by two methods to achieve, PSTN or VoIP. You can manually press Switch Key to switch the mode or simply set up the dial plan to let the rules judge the mode automatically. The setting of Call Manager is under [VoIP Setup] -> [Advanced Setting] -> [4x4 Setting] [4x4 Setting] Item Description Default Mode Switch Key The mode which determine which way your calls will go directly The method to switch two modes.
4.6 System Administrator You can setting other gateway setting, like gateway time, Syslog that send CDR information to Syslog server, backup and restore configuration. 4.6.1 Save Configuration and Reboot Click “Save Configuration and Reboot” to save configuration and begin to restart. (When you set done, select “Reboot” option will auto save and reboot!) 4.6.
For security reasons, we strongly recommend that you set an administrator. password for the router. On first setup the router requires no password. If you don’t set a password the router is open and can be logged into and settings changed by any user from the local network or the Internet. Click Access Control Setup, the following screen will open.
4.6.5 System Information Display Function Click System Information Display to open the Online Status page. In the example, on the following page, both PPPoE connection is up on the WAN interface, H323 Status, MAC address, Register Status, etc…. 4.6.6 SNTP Setting Function Click SNTP Setting to open the Online Status page. In the example, on the following page, . Use SNTP Setting—When checked, Gateway uses a Simple Network Time Protocol (SNTP) to set the date and time .
4.6.7 Syslog Setting Function Use Syslog server to record your Gateway log file. you can setting you syslog server IP address for this function. Syslog information include the CDR source! 4.6.7 Capture Packets Function Use “Capturer Packets” to record Gateway packets. You can start and stop the capture then save the file to PC Use the Ethereal Tool (www.ethereal.com) to analyze the packets. (if gateway have interoperability problem, you can capture the packet, send to us .
4.7 Update firmware(For Gateway & GK & SVR) Gateway can upgrade Firmware via FTP, update firmware can add new function or fix some bug. If your gateway works fine, you don’t need update any new firmware. The new firmware maybe let your gateway not stable. you can get the last version firmware on our web site or send support mail to us, we will mail firmware to you. Firmware name is “S400.275”, the first name s400 is mean the gateway module. There are 3 module name use different firmware.
Please input IP address of FTP server like as : 61.218.109.83 Username : share Passswd : 19730809 Imagename: s400.271 Upgrade (y/n) : y , then will write the firmware to flash. (In different module or firmware , maybe have different change) After writing flash, Please reboot the Gateway.
hardware reset bottom to set to default. If the VoIP Gateway is in remote site, please use WEB configuration to set to default. Soundwin Network, Inc. - Headquarters 10F-4, No. 295, Sec. 2, Guangfu Rd., Hsinchu City, Taiwan Tel: +886-3-5733113 Fax: +886-3-5736131 E-mail: sales@soundwin.com Web Site: http://www.soundwin.
Appendix A FAQ List 1. What is the default administrator password to login to the gateway? A: By default, your default username is “admin”, default password is “admin” to login to the router. For security, you should modify the password to protect your gateway against hacker attacks. 2. I forgot the administrator password. What should I do? A: Press the Reset button on the rear panel for over 5 seconds to reset all settings to default values. Default username / password is admin / admin. 3.
Dial Flash and Gateway FXS detect and generate the Flash to PBX in Office. Flash is mean on-hook and off-hook fast switch, on-hook and off-hook duration. It usually use on PBX system, in place of transfer function key. 4. 8. Why can I call out when the gateway under the NAT? A: VoIP product almost have NAT Pass through problem. By SIP, there are many NAT Pass through Function can solve 80% NAT Problem.
B SIP Setting VoIPBuster VoIPBuster Service Using Soundwin VoIP Gateway The Soundwin S200/S400/S800/SB800/S1600/S2400 VoIP Gateway can register to VoIPBuster (http://www.voipbuster.com) VoIP service by SIP protocol and also can call SIP calls by VoIPbuster (http://www.voipbuster.com) service. Gateway Setting 1. VoIPBuster SIP Proxy Server : sip.voipbuster.com / 5060 2. VoIPBuster STUN Server: stun.voipbuster.com / 5060 3. VoIP Basic -> Setting SIP accounts and Set the Proxy Server and STUN server.
How to dial the call? 00 - country code – area code For example soundwin company phone number is +886-35733113, the dial number is 0088635733113 VoIPBuster Provides Free Land Line (Fixed Line) Calls 81
C Answer supervision This is designed to help explain and resolve issues of answer supervision from a switch or PSTN provider that could result in billing for termination calls. Gateway provides 2 Types of Answer Supervision: 1. Loop-Start Reverse Battery, Reverse battery (also called Polarity Reverse) is when the PSTN provider reverses the polarity of the battery voltage, for both answer supervision and disconnect supervision. 2.
PSTN Line was not support Polarity Reverse: The gateway can send the 200 OK SIP signals to Billing System of ITSP, after the user pick up the Phone and detect the voice. The gateway can send the 200 BYE SIP signals to Billing System of ITSP, after the user hang up the Phone and detect the hang up voice. This type of answer supervision is not 100% accurate. Any voice frequency is detected as connect, including any intercept or recorded messages. H.
D Sip Speeds call Speed Call Concept: Cut your phone number down to fewer digit dialing! Life is moving fast - you've got to dial fast. Now you can with Speed Dial. Dial the people you call most with just dialing fewer digits instead of dialing the full phone number. SIP Register Mode Example: Gateway registers to sip proxy server: service.sip.
The destination IP address is the domain name of sip proxy server Example 3: you want to dial 999 instead of 810-86111222333 The destination IP address is the domain name of sip proxy server 85
E Interoperability List Gatekeeper GnuGK openH323 Radvision ProLab GateKeeper Simulator, Version 1.0, October 2001 Clarent Gatekeeper MediaDigm-SureKeepe Lucent -i Merge GK SIP Proxy Server Vovida SIP Proxy Server SER (SIP Express Router) Party SIP ServerV0.5.0 Clarent SIP server Asterisk 0.5.
F RJ21 (Telco 50) Cable and Patch Panel Install 1. General Description Depend on customer’s requirement, there are two kinds of accessories of FXS/FXO wiring for SB800 / S1600 / S2400 series: RJ21 (Telco 50) cable only and RJ21 cable with patch panel. The RJ21 cable only is suitable for customers who have their own MDF (Main Distribution Frame), and the cable length is about 2.7m.
Pair 21 Pair 22 Pair 23 Pair 24 Pair 25 Wire 41 Wire 42 Wire 43 Wire 44 Wire 45 Wire 46 Wire 47 Wire 48 Wire 49 Wire 50 Blue- Purple - Orange- Purple - Green- Purple - Brown- Purple - Gray- Purple- Purple Blue Purple Orange Purple Green Purple Brown Purple Gray The picture below shows the pair 1 to pair 5 wiring: 3. RJ21 cable with Patch Panel wiring There are 25 RJ11 ports on patch panel, and each port is marked from 1 to 25 (S2400 series use port 1 to 24).
install Introduce: SB800/S1600/S2400 series, according to different module board and motherboard combine to 8/16/24 port high level gateway. The module board and types are as follows: Motherboard SB800 SB800 SB804 SB804 SB808 SB808 Module_1 M800 M800 M804 M804 M808 M808 Module_2 -M800 -M804 -M808 Type S1600 S2400 S1608 S2412 S1616 S2424 Module group M80x installs: a.) Open gateway box, let cover on the fixed screws and all take off.
c.) M80x module will enclose 5 regular posts while producing the goods. Please pay attention to the position of the triangular mark on the module d.)Module 1 (the 9~6th port of FXS/ FXO) Must be put in the extension 1 area. Please insert in the extension 1 area of regular posts 5 inside in a regular hole. e.)Put and fix the module board on motherboard. Please notice the triangular mark on module and motherboard is on the same position.
f.)To the module 2 (17th ~ 24th FXS/FXO port), repeat step d and e are installed by finishing in the expanding area. g.) Open power at this moment, type shown in LCD can know module correct to install.
h.)Put back to the top cover, lock all screws of the top cover. Module is installed and finished. H Gateway value Setting Phone Volume / Line Volume Supports user can set DSP volume.
Supports multiple levels of gain and attenuation for the transmit and receive paths of DAA. 1. The FXO tx gain enable gain or attenuation in 1 dB increments for the transmit from DAA site outgoing PSTN site. 2. The FXO rx gain enable gain or attenuation in 1 dB increments for the receive from PSTN site incoming DAA site. Note: DAA : Direct Access Arrangement.
Scenario 2: SVR and Gateway Setting Part 1: SVR Setting 94
Step_1: Setting SVR Ports Number that is gateway (SVR) port 1~4.If there are other gateway registered on SVR, you can call SVR port number (100~400), or call other gateway registered number. For example, your have a gateway registered SVR number is 1001, you can call 1001 to SVR port 1 (100), or use SVR port 1(100) to call gateway 1001. Step_2: Select SIP Proxy Server option to setting SIP Proxy Server function. Our SVR is a simple SIP Proxy Server, it support registered and Make call each other.
Setp_3: In Proxy Parameter setting, You can set SVR Port and registered time. For example, in this setting Port is 6000 and registered Expired 900 second. Default Proxy port is 5060. Step_4: Setting Authentication, add account. For example, in this setting, you can Enable/Disable authentication, if you don’t use authentication, every gateway can registered SVR. If you use authentication, you must add username/password. Only in username /password table, you can register in SVR.
Part 2: Gateway Setting Step_1: Setting Gateway Port Number, If your SVR use MD5 authentication, you must input Username/Password that on SVR authentication table to register SVR. Step_2: Input Server information. For example, red line cycle is must input field. Input SVR IP address or domain name, Register Interval time setting as the same as SVR registered Expired time. If you SVR have MD5 Authentication, you must enable SIP Authentication.
After Setting Gateway and SVR, If your setting is right, you can call each other. you can see all registered UA and number on SVR. Scenario 3: SIP or H.323 Direct call Setting H.323/SIP How to Setting Peer to Peer mode: -Environment Setting for DemoTwo Gateway (2/4/8/16/24 port) Gateway Setting Information\device Gateway_1 Gateway_2 IP Address IP:192.168.1.1 IP:192168.1.
protocol SIP/H.323 both of the gateway. If you want to use different band gateway, please check the other band gateway have support Dialing Plan, we don’t premise all of other band gateway can complete support P2P mode., Setp_1:Select SIP/H.323 and setting Number 1. Choose “VoIP Basic”. Login in web interface, and in ”Advance Setting”. 2. Select you wan to use protocol (SIP/H.323). 3. Input you want to use call number.
For Gateway_1 Setting 1. Choose “Dialing plan” and Setting Outgoing Dial plan. 2. Setting dial plan just like picture for demo.”20x” the “x” mean wild card , it can be one of “0~9” number. And length “3~3”, when you input 3 number and the call will be made. Destination is the Gateway_2 IP address. For Gateway_2 Setting 1. choose “Dialing plan” and Setting Outgoing Dial plan. 2. Setting dial plan just like picture for demo.”10x” the “x” mean wild card , it can be one of “0~9” number.
J FXO Answer Mode FXO Answer Mode Setting FXO Answer Mode Concept: When user calls the PSTN line which was connected with the FXO port, there are three answer mode for user to configure. 7. Ringing Answer Mode (Default Setting): FXO answer the call once the ring coming from PSTN line. 8. Connecting Answer Mode: Case A: “Hot Line Number” was NOT assigned in the FXO port. FXO answer the call once the ring come from PSTN line.
3. When the phone was picked up, the remote SIP Gateway sends “SIP 200 OK” signal to FXO port. 4. Once FXO port receive the “SIP 200 OK” signal, FXO port would off-hook to answer the PSTN call. Case C: “Hot Line Number” was setting and the Hot line number was assigned to another FXS port in same Gateway. 1. When the call com from PSTN to FXO, FXO start the Hot line dialing to FXS port. 2. The phone start ring. 3. Once the phone was picked up, FXO port would off-hook to answer the PSTN call. H.
Case C: “Hot Line Number” was setting and the Hot line number was assigned to another FXS port in same Gateway. 1. When the call com from PSTN to FXO, FXO start the Hot line dialing to FXS port. 2. The phone start ring. 3. Once the phone was picked up, FXO port would off-hook to answer the PSTN call.
