MicroNet Technology SP5001A-S User Manual

Download

User’s Manual

IP Telephony Gateway

Model No.: SP5001A/S

http://www.micronet.info

Table of Contents

1. INTRODUCTION..................................................................................................................................4

1.1. OVERVIEW.....................................................................................................................................4

1.2. FEATURES .....................................................................................................................................5

1.3. DEFAULT SETTINGS........................................................................................................................ 6

1.4. APPEARANCE................................................................................................................................. 7

2. SETTING UP THE GATEWAY.............................................................................................................9

2.1. CONNECTING THE SP5001A/S .......................................................................................................9

2.2. INTERNET CONNECTION SETUP.....................................................................................................13

2.3. PROXY MODE SETUP ...................................................................................................................16

2.4. PEER-TO-PEER MODE SETUP.......................................................................................................18

2.5. P2P CONNECTION EXAMPLE ........................................................................................................ 21

3. ADVANCED SETUP..........................................................................................................................23

3.1. BEHINDS THE NAT ROUTER (P2P MODE)...................................................................................... 23

Setup the SP5001A/S: ...................................................................................................... 24

Setup the NAT router: .......................................................................................................26

3.2. CODEC SELECTION ......................................................................................................................28

4. FIRMWARE UPGRADE.....................................................................................................................30

4.1. TFTP SERVER SETUP..................................................................................................................31

4.2. UPGRADE BY WEB INTERFACE ..................................................................................................... 34

4.3. UPGRADE BY TELNET COMMAND...................................................................................................37

5. WEB CONFIGURATION MENU........................................................................................................41

5.1. NETWORK INTERFACE ..................................................................................................................42

5.2. SIP INFORMATION ........................................................................................................................ 45

5.3. SYSTEM CONFIGURATION .............................................................................................................47

5.4. PPPOE CONFIGURATION..............................................................................................................48

5.5. VOICE CONFIGURATION ................................................................................................................49

5.6. PHONE PATTERN CONFIGURATION ................................................................................................ 51

5.7. SUPPORT CONFIGURATION ...........................................................................................................53

5.8. PHONE BOOK CONFIGURATION ..................................................................................................... 54

5.9. PREFIX CONFIGURATION...............................................................................................................55

5.10. DSCP CONFIGURATION................................................................................................................ 56

5.11. PASSWORD CONFIGURATION ........................................................................................................58

5.12. ROM UPGRADE ...........................................................................................................................59

5.13. FLASH CLEAN ..............................................................................................................................60

5.14. COMMIT DATA .............................................................................................................................. 60

5.15. REBOOT SYSTEM .........................................................................................................................60

6. COMMAND LIST................................................................................................................................61

6.1. [HELP] .........................................................................................................................................62

6.2. [QUIT] .......................................................................................................................................... 62

6.3. [DEBUG]....................................................................................................................................... 63

6.4. [REBOOT]..................................................................................................................................... 64

6.5. [COMMIT] .....................................................................................................................................64

6.6. [IFADDR] ......................................................................................................................................65

6.7. [TIME] ..........................................................................................................................................68

6.8. [PING]..........................................................................................................................................68

6.9. [PBOOK].......................................................................................................................................69

6.10. [PPPOE] .......................................................................................................................................71

6.11. [FLASH]........................................................................................................................................72

6.12. [SYSCONF]...................................................................................................................................73

6.13. [SIP] ............................................................................................................................................75

6.14. [SECURITY] ..................................................................................................................................76

6.15. [VOICE] ........................................................................................................................................77

6.16. [SUPPORT] ...................................................................................................................................79

6.17. [TOS] ...........................................................................................................................................80

6.18. [PHONE]....................................................................................................................................... 81

6.19. [BUREAU].....................................................................................................................................83

6.20. [ROM] ..........................................................................................................................................84

6.21. [PASSWD].....................................................................................................................................85

6.22. [PREFIX].......................................................................................................................................86

About this User’s Manual

This user’s guide gives hardware specifications and explains web configuration and command line configuration for the VoIP Telephony Gateway.

Specifications are subject to change without notice.

Online Upgrade

Please refer to

General Syntax Conventions

Mouse action sequences are denoted using a comma. For example, click start, Settings, Control Panel, Network means first you click Start, Click or move the mouse pointer over Settings the click or move the mouse pointer over Control Panel and finally click (or double-click) Network. “Enter” means to type one or more characters.

Predefined choices are in Bold Arial Font. A single keystroke is in Arial font and enclosed in square brackets. [Enter] means the

Enter. For brevity’s sake, we will use “e.g.,” as shorthand for “for instance”, and “i.e.,” for “that is” or “in other words.”

Safety Notes

http://www.micronet.info/ for additional support documentation.

Use the external power supply that is included in the package. Other power supplies may cause damage to the phone, affect the behavior or induce noise.

1. Introduction

1.1. Overview

Micronet SP5001A/S FXS Gateway is designed to connect standard telephone devices to IP-based telephony networks, providing users with high-quality VoIP service. In addition, the 10/100M switch ports can offer network connection to co-located PC or other Ethernet-based devices. No need to prepare extra hubs or switches. SP5001A/S is an ideal solution for home users or small offices.

SP5001A/S is compliant with IETF RFC 3261 SIP standards, and has a built-in DHCP server to assign IP addresses automatically to your PCs, making configuration effortless. SP5001A/S can save the toll call expense and maximizes your broadband investment.

1.2. Features

● Compliant with IETF RFC 3261 SIP standards

● Provide 1 RJ-11 FXS port for phone set or fax machine

● Provide 3-port 10/100M Ethernet switch

● Provide advanced telephony features, such as call hold, call forward and call

transfer.

● Support Proxy and Peer-to-Peer Mode

● Support FAX over IP (T.38)

● Support FSK and DTMF Caller ID

● Support Static IP, DHCP and PPPoE connection

● Built-in NAT for IP sharing

● Built-in DHCP Server

● TFTP/FTP firmware upgrade

● QoS : ToS (Type of Service)

● Support EMS (Element Management System)**

Audio feature

● Codec: G.711 a/µ-law, G.723.1 (6.3kbps), G.729A

● VAD (Voice Activity Detection)

● CNG (Comfort Noise Generate)

● G.168/165-compliant adaptive echo cancellation

● Dynamic Jitter Buffer

● Bad Frame Interpolation

● Voice/DTMF Gain Settings

Interface

● One 10/100 Base-T Ethernet RJ45 port for WAN

● Three 10/100 Base-T Ethernet RJ45 ports for LAN

● One RJ11 Telephone Port (FXS).

● DC 12V input.

System Management

WEB Interface, Telnet

Environment

Operating and storage Humidity: 10 to 90 % (Non-condensing) Operational Temperature: 0 to 40 ℃ Storage Temperature: -10 to 50 ℃

Dimension & Weight : 190 x 124 x 37 mm, 320g

Certification

CE, FCC

1.3. Default Settings

The following are the settings of the default profile

IP Parameters

WAN IP Address: 10.1.1.3 Subnet: 255.0.0.0 Default gateway: 10.1.1.254 LAN IP Address: 192.168.123.123

Telnet and Web Login Password

1.4. Appearance

Front Panel

1. LED Status Display

Rear Panel

1. Reset Button

2. RJ-45 WAN Port

3. RJ-45 LAN Ports

4. RJ-11 FXS Interface

5. Power Jack 12V DC

Note :

To restore the factory default configuration settings, press and hold the Reset button on the rear panel for more than 3 seconds. Release the Reset button and wait for the gateway to reboot.

LED Status Display:

LEDs Functions Status Active Description

PWR

STATUS

WAN

Power Green

Status Green

TEL Green

WAN Green

On The Power is on Off The Power is off

Off Gateway is in Peer-to-Peer Mode

Blinking

On The Telephone is Off-Hook

Off The Telephone is On-Hook TEL

Blinking The gateway has Incoming Call On WAN Port connected Off WAN Port disconnected

Blinking

On LAN Port connected

Gateway is under Proxy mode and registered to Proxy server successfully

Gateway is in Proxy mode but no register, or Gateway is booting up

WAN Port is transmitting or receiving data

LAN

(1, 2, 3)

Ethernet WAN Port:

Connect the Ethernet cable from gateway's WAN port to the ADSL or Cable modem Ethernet port.

Ethernet LAN ports:

Connect the Ethernet cable from gateway's LAN port to the Ethernet adapter in your computer.

TEL Port:

RJ-11 connector, FXS interface. To connect analog phone set or trunk line of PABX.

Power Jack:

12V DC Power supply.

Connection

Green

Off LAN Port disconnected LAN

LAN Ports are transmitting or

Blinking

receiving data

2. Setting Up the Gateway

This chapter will describe the basic connection setup and configure the gateway via your web browser through a computer. It outlines how to connect your VoIP Gateway to the LAN and the WAN. In the case of connecting a Cable Modem you must connect the coaxial cable from your cable service to the threaded coaxial cable connect on the back of the cable modem.

2.1. Connecting the SP5001A/S

1. Turn off your computer

2. Turn off the DSL or cable broadband modem

3. Connect the Ethernet cable from WAN port to the ADSL or Cable modem Ethernet port.

4. Connect the Ethernet cable from LAN port to the Ethernet adapter in your computer

5. Connect the telephone handset to the TEL port (FXS port)

6. Connect the power adapter to the gateway and plug it in to a power outlet. It takes about 40 seconds to boot up completely

7. Power on your computer and DSL modem

8. Configure your PC network adapter to set to automatically get its TCP/IP configuration from the SP5001A/S via DHCP.

SP5001A/S provides DHCP server function, the Dynamic Host Configuration Protocol is a communications protocol that lets automate the assignment of Internet Protocol (IP) addresses in an organization's network.

9. Connect the SP5001A/S by typing http://192.168.123.123 in the address field of Internet Explorer or Netscape Navigator.

10. Login your gateway by enter root as user name and no password when

prompted.

If network failed to access the Internet, restart your network in the correct sequence. Failure to restart your network in the correct sequence could prevent you from connecting to the Internet.

1. First, plug in and turn on the broadband modem and wait 1 or 2 minutes.

2. Second, plug in the power to your VoIP gateway and wait 1 minute.

3. Last, turn on your computer.

2.2. Internet Connection Setup

This section shows the basic setup to enter the Internet connection settings provided by your ISP. Before proceeding with the Internet connection setup, you need to know the setup information for your specific type of Internet connection, for example, DSL connection or Cable connection, login name / e-mail and password, then you can configure the gateway.

A. PPPoE Connection Setup

For DSL users, many ISPs may require you to log on with a user name (or e-mail address) and password to gain access to the Internet. This connection type is called Point to Point Protocol over Ethernet (PPPoE). PPPoE (Point-to-Point Protocol over Ethernet) is a specification for connecting multiple computer users on an Ethernet local area network to a remote site through common customer premises equipment, which is the telephone company's term for a modem and similar devices, commonly used in dialup connections, users share a Digital Subscriber Line (DSL), cable modem, or wireless connection to the Internet. Most of the PPPoE connection is temporarily assigning an IP address to a requesting Dynamic Host Configuration Protocol (DHCP) NAT router or computer from a pool of IP addresses. The temporary IP address is

called a dynamic IP address.

1. Select the [PPPoE Configuration]

2. Select [On] to enable the PPPoE Device

3. Enter your DSL login name into User Name field

4. Enter your DSL password into Password field

5. Click

6. Select [Commit Data] and click button.

7. Select [Reboot System] and click button.

Wait for the gateway to reboot, check the DSL connection status by select the [PPPoE Configuration]

button

B. Static DSL Connection Setup

A static IP address is a number (in the form of a dotted quad) that is assigned by an Internet service provider (ISP) to be its permanent address on the Internet. VoIP gateways use IP addresses to locate and talk to each other on the Internet, much the same way people use phone numbers to locate and talk to one another on the telephone.

1. Select the [Network Interface]

2. Enter the IP address, Subnet and Default Gateway

3. Click

button

4. Select [Commit Data] and click button.

5. Select [Reboot System] and click button.

2.3. Proxy Mode Setup

You can choose either Proxy mode or Peer-to-Peer mode for communication. Proxy mode requires account information to access to the service; it’s assigned by the SIP service provider. SIP Serve rendezvous point at which callees are globally reachable, and perform registration, call routing function. The VoIP gateway (or IP phone) of the users in the domain register their IP addresses with the server so that the other users can reach them. Proxy Mode also suit for the gateway has dynamic IP address connection.

1. Select the [SIP Information] at the Configuration Menu section

2. Enter the Proxy Server’s IP address or URL. For example, 220.130.173.70 or sip.micronet.info

3. Enter the Line Number

4. Enter the Account. It can be same as the Line number, the user name or the e-mail account. Check with your VoIP service support for the details.

5. Enter the Password

6. Click

button

7. Select [Commit Data] and click

button.

8. Select [Reboot System] and click button.

After reboot the SP5001A/S, check the Status LED, it shows the gateway has registered to the SIP server successfully when the LED stays on. If not, the Status

LED is blinking, check the Internet connection and SIP Configuration settings again.

2.4. Peer-to-Peer Mode Setup

P2P Mode doesn’t require any centralized control units like Proxy Mode does, it makes

communication between two end-points directly, [Phone Book] needs to configure to

work with in P2P Mode. It requires direct public IP access, it can also perform the job behind the NAT device with static public IP connection, but it can not work behind the NAT device with the dynamic IP connection.

1. Select the [SIP Information] at the Configuration Menu section

2. Enter the Line Number

The Line Number is same as an extension number from of PABX system. You can create your own extension numbering plan for your VoIP system.

3. Click

button

4. Select the [Phone Book] at the Configuration Menu section

5. Enter the destination information into New Record field

6. Select [Commit Data] and click

button.

7. Select [Reboot System] and click button.

Follow the same steps to configure the remote side gateway. The next section shows the Peer-to-Peer configuration example.

Note:

Remember, the P2P Mode can not work behind the NAT device with the dynamic IP connection.

2.5. P2P Connection Example

The following example shows the gateway’s settings of each location in P2P Mode

Head office dials 77# to reach branch office gateway Branch office dials 33# to reach head office gateway

Head Office

Network

SIP Line Number

Phone Book

Branch Office

Network

SIP Line Number

Phone Book

3. Advanced Setup

It would be simple if every VoIP gateway, or computer that connects to the Internet could have its own static IP number, but when the Internet was first conceived, the architects didn't foresee the need for an unlimited number of IP addresses. Consequently, there are not enough IP numbers to go around and we use the NAT device or router connects our local area network (LAN), or the group of PCs in your home or office, to the Internet. In this section, we will show you how to configure your SP5001A/S behind the NAT device if your SP5001A/S acts standalone device in your network.

3.1. Behinds the NAT Router (P2P Mode)

When you place the SP5001A/S behinds the NAT router in P2P Mode, a few more settings need to configure on the SP5001A/S and router.

Setup the SP5001A/S:

1. Open the web browser to connect the gateway and select the [Network Interface]

2. Change the WAN IP Address, assign the IP address depends on your router

settings, for example: 192.168.1.8

Note: The static IP must configured in this application

3. Change the Subnet Mask if necessary, for example: 255.255.255.0

4. Change the WAN IP Address. Here means your router’s LAN IP address, for

example: 192.168.1.1

5. Select the [IP Sharing] and enable

6. Enter the IP Sharing Server Address, here means your router’s WAN IP address, for example: 61.219.198.204

7. Click the

button

8. Select [Commit Data] and click button.

9. Select [Reboot System] and click button.

Setup the NAT router:

When the VoIP gateway or computer behind the NAT router, normal web surfing and email will not know the difference, but some communication services using ports other then the normal web ports (ports are like door ways to your computer and for security the communication has the abnormal ones closed and locked but the normal ones

open like the web surfing port 80). It must enable the DMZ (Demilitarized Zone) function or setup the Port Forwarding(or called Virtual Server), let the communication

traffic can pass through the router.

A. DMZ Setup

1. Enter the NAT router configuration by web browser or software utility.

2. Locate the DMZ function and enable it

SP888B Broadband Router Screen Shot

DMZ

In computer networks, a DMZ (demilitarized zone) is a computer host or small network inserted as a "neutral zone" between a company's private network and the outside public network. It prevents outside users from getting direct access to a server that has company data. (The term comes from the geographic buffer zone that was set up between North Korea and South Korea following the UN "police action" in the early 1950s.)

B. Port Forwarding (Virtual Server)

If you can not find the DMZ function on your router or firewall, then the Port Forwarding is another way to allow the communication traffic pass through. A broadband router creates a firewall between your internal network and the internet. A firewall keeps unwanted traffic from the internet away from your LAN computers. A ‘tunnel’ can be created through your firewall so that the computers on the Internet can communicate to one of the computers on your LAN on a single port. This is handy for running web servers, game servers, ftp servers, VoIP applications or even video conferencing. This is called port forwarding. Port 5060 and Port 16384 ~ are commonly used in SIP from Micronet VoIP products

1. Enter the NAT router configuration by web browser or software utility.

2. Locate the Port Forwarding function and enable it

SP888B Broadband Router Screen Shot

Port

Applications running on TCP/IP open connections to other computers using something called ports. Ports allow multiple applications to reside on a single computer - all talking TCP/IP. Ports are another set of numbers AFTER the standard IP address. Applications often hide these port numbers to reduce the complexity of TCP/IP. Example: web services (HTTP) reside on port 80 by default, port 5060 is for the SIP signaling by default, port 16384 ~ is for the RTP by default, etc.

3.2. Codec Selection

Codec (Coder / Decoder)

Codecs are used to convert analog signals to a digital bit stream, and another identical codec at the far end of the communication converts digital bit stream back into an analog signal. Codecs vary in the sound quality, the bandwidth required, the computational requirements, etc. Codecs generally provide a compression capability to save network bandwidth. Some codecs also support silence suppression, where silence is not encoded or transmitted. In the VoIP world, codec's are used to encode voice for transmission across IP networks. Micronet VoIP gateway supports several different codecs, G.711A/µ law, G723.1, G729, and when talking to each other, negotiate which codec they will use.

Codec Description

G.723.1 G.723.1 is an ITU-T

standard codec. Its reasonably low bit rate (6.3Kbps or 5.3Kbps). Use of this codec in a product requires licensing by Sipro Lab Telecom

G.729 Coding of speech at 8 kbit/s

using conjugate-structure algebraic-code-excited linear-prediction (CS-ACELP)

G.711 G.711 is the international

standard for encoding telephone audio on an 64

Bit Rate

Remark

(Kb/s)

5.6 / 6.3 It encodes speech or other audio signals in frames using linear predictive analysis-by-synthesis coding.

8 Low delay (15 ms)

64 mu-law (US, Japan) and A-law

(Europe) companding

kbps channel. It is a pulse code modulation (PCM). This is most

All VoIP packets are made up of two components: voice samples and IP/UDP/RTP headers. Although the voice samples are compressed by the Digital Signal Processor

(DSP) and may vary in size based on the codec used, these headers are a constant 40 bytes in length. This table shows the nominal Ethernet bandwidth consumption.

Codec

Nominal Ethernet Bandwidth

(Kb/s)

20.8 kbps (for 5.3 frame bit rate)

G.723.1 5.3 / 6.4

21.9 kbps (for 6.4 frame bit rate)

G.729 8 31.2 kbps

G.711 64 87.2 kbps

Bit Rate

4. Firmware Upgrade

Firmware is a combination of software and hardware. Computer chips that have data or programs recorded on them are firmware. These chips commonly include the

following: ROMs (read-only memory), PROMs (programmable read-only memory), EPROMs (erasable programmable read-only memory), it’s same as software, except it

is executed from ROM, and does not disappear when the power is turned off. Firmware in PROM or EPROM is designed to be updated if necessary through a software update. You can download firmware updates for Micronet VoIP products from Micronet web site at Download Center.

You must have a TFTP or FTP server configured and running to perform the download operation, you can download the TFTP program from Micronet web site.

Note:

Firmware should be upgraded ONLY if you experience problems with the Gateway

4.1. TFTP Server Setup

1. Click the link to download the TFTP program. The file downloads as a compressed file called a zip (.zip) file.

http://www.micronet.info/Download/Driver/VoIP/utility/tftpd32m.zip

Note:

The file opens in WinZip® or another decompression program, use the program to extract the zip from the compressed file.

2. Select the folder where you want to save the compressed file, and then click Save

3. After extracting the file, you can double-click the tftpd32.exe to start the

program.

4. Download the firmware, copy and decompress the file into same folder where the TFTP program located

Firmware file name: 1asipfxs.107b

5. Now, you have the TFTP server and latest Application firmware ready. Go to next section to configure the SP5001A/S.

Note:

Make sure the Current Directory has located the same folder as where the firmware

file saved.

4.2. Upgrade by WEB Interface

Assumed your SP5001A/S has configured static IP address (192.168.1.8) as the

diagram showed.

1. Open the web browser to connect the gateway and select the [ROM Configuration]

2. Enter the TFTP server IP address (192.168.1.12)

3. Enter the firmware file name into Target File name field

4. Select [TFTP] method

5. Select [Application Image] as Target File Type

6. Click button to start downloading the firmware

TFTP server is uploading the file to the gateway

After file transferred complete, the gateway will write the new firmware into the Flash

ROM, wait until see the web browser showed Please issue FLASH CLEAN to consist software version

7. Select [Flash Clean] menu and click the button

When the screen showed Flash cleaned!! You can now reboot the gateway.

Note:

All the settings will be erased after upgrade the firmware, the gateway needs to re-configure again.

You can change Micronet VoIP gateway’s protocol from SIP to H.323, or from H.323 to SIP by firmware uploaded as well.

4.3. Upgrade by Telnet Command

Telnet is another way you can access the gateway, assuming it has given you permission. More technically, Telnet is a user command and an underlying TCP/IP protocol for accessing remote network devices. On the Web, HTTP and FTP protocols allow you to request specific files from remote network devices, but not to actually be logged on as a user of that computer. With Telnet, you log on as a regular user with whatever privileges you may have been granted to the specific application and data on that device. Telnet is most likely to be used by program developers and anyone who has a need to use specific applications or data located at a particular host.

A Telnet command request looks like this: telnet 192.168.1.8

We will show you how to use the telnet command to do the upgrade firmware under Windows OS. For the Linux OS users, please check your OS manual or ask the network administrator for the help.

1. Click Start and select Run…

2. Type telnet 192.168.1.8 and click to connect your gateway

3. Login screen

4. Type root as login name, no password, press [Enter]

5. Type the following command line to execute the upgrade procedure.

usr/config$ rom -app -s 192.168.1.12 -f 1asipfxs.107b

6. Gateway is downloading the firmware file

7. After downloaded the file, start writing into Flash ROM

8. Always clean the flash memory after upgraded new firmware.

9. After the rebooting ... message showed, close the Telnet windows

10. Re-login the gateway and configure it.

Note:

Telnet mode is good for user to monitoring the upgrade procedures

For more details about the Telnet commands, refer to the Command List section

5. WEB Configuration Menu

Micronet gateway provides a built-in web server. You can be accessed via Microsoft Internet Explorer or Netscape Navigator through use of a computer connected with an Ethernet cable to the SP5001A/S gateway. Configuration and administration can be performed through this convenient web interface. This section shows all of the configure functions.

Main page

5.1. Network Interface

LAN IP Address

WAN IP Address

Subnet Mask

When the gateway connects to the DSL modem directly and shares the connection for the LAN devices, this IP Address is assigned for the LAN devices’ default gateway

If the gateway acts stand alone device and behinds the NAT router, the LAN IP Address can be leave as it is and disable the NAT function

When your gateway has static IP address or set behind the NAT

router, configure the WAN IP Address, Subnet Mask and Default routing gateway together

A subnet is an identifiably separate part of an organization's network. Typically, a subnet may represent all the machines at one geographic location, in one building, or on the same local area

network (LAN). The appropriate subnet mask carried along with the

packet would be: 255.255.255.0

Default routing gateway

DHCP

NAT

SNTP

SNTP Server Address

The default gateway IP is assigned by the ISP or your NAT router’s LAN IP.

Enable the DHCP client function if you have the Cable Modem connection to access the Internet

Network Address Translation. NAT located where the LAN meets the Internet makes all necessary IP address translations. Enable this function if your gateway is sharing the Internet connection with PC, notebook or some other network devices.

Simple Network Time Protocol. It’s a simplified version of the NTP protocol, it is an Internet protocol used to synchronize the clocks of gateway to some time reference

Enter the preferred Time server address here You can find the public SNTP server list on Microsoft web site

GMT

IP Sharing

UPnP

http://support.microsoft.com/default.aspx?scid=kb;EN-US;q262680

SNTP uses UTC(Universal Time Coordinated) as reference time, formerly and still widely called Greenwich Mean Time (GMT). Set the correct GMT for your location to get time display correctly.

Enable this function when you place the gateway behind the NAT router device.

Universal Plug and Play (UPnP) is a standard that uses Internet and Web protocols to enable devices such as PCs, peripherals, intelligent appliances, and wireless devices to be plugged into a network and automatically know about each other. With UPnP, when a user plugs a device into the network, the device will configure itself, acquire a TCP/IP address, and use a discovery protocol based on the Internet's Hypertext Transfer Protocol (HTTP) to announce its presence on the network to other devices. If the gateway behinds the NAT router with UPnP supported, you

can enable the UPnP function.

IP Sharing Server Address

Primary DNS Server

Secondary DNS Server

Enter the public Internet IP address here, if the gateway behinds the NAT router.

Enter the primary Domain Name Server IP address here. If the gateway connects to the server by URL address, DNS must configure. DNS is the Domain Name System. DNS converts machine names to the IP addresses that all machines on the net have. It translates from name to address and from address to name. For example

www.micronet.info

Enter the secondary Domain Name Server IP address here.

5.2. SIP Information

Run Mode

Primary Proxy IP Address

Secondary Proxy IP Address Outbound Proxy

Select Proxy mode or Peer-to-Peer mode.

Set Proxy IP Address or URL e.g. 220.130.173.70 or sip.micronet.info

Set secondary proxy address here if available

The outbound proxy is a normal SIP proxy. You configure your client, the gateway or phone, to use the proxy for all SIP sessions, just like when you configure your Web browser to use a Web proxy for all Web transactions. In some cases, the outbound proxy is placed alongside the firewall and is the only way to let SIP traffic pass from the internal network to the Internet. Enter the Outbound Proxy address here if your SIP service provider supported.

Proxy Port

SIP local UDP port number (5060~5070), default: 5060. Change the Proxy port only when your service provider has different application.

Prefix String

Line1 Number

Line1 Account

Line1 Password

SIP Port RTP Port

The Line Number is same as the telephone number, people locate you by this number.

Account is requires by the SIP server for register, it can be the Line number, user name or e-mail account.

Enter the Account password here.

SIP Signaling port Real-time Transport Protocol. The Internet protocol for transmitting real-time data such as audio and video. RTP itself does not guarantee real-time delivery of data, but it does provide mechanisms for the sending and receiving applications to support streaming data.

Expire

Set expire time to match the SIP server registration time required. It means, if you set 60, the gateway sends the register request information to the SIP server every 60 seconds.

5.3. System Configuration

Keypad Type

RFC2833 Payload Type

FAX Payload Type

Inter Digit Time

CallerID Type

Select In-Band, RFC2833 on DTMF replay type

RFC2833 Payload Type (range: 96~128 inter-used: 100, 102~105)

Set Fax Payload Type (range: 96 or 101, default: 101)

Set the DTMF inter digit time (second)

Set CallerID type. If your telephone set has CallerID function, after the first ring at destination site, device will send line number as Caller ID to called site. FSK (Frequency-shift keying) is a method of transmitting digital signals. DTMF (dual tone multi frequency) is the signal to the phone company that you generate when you press an ordinary telephone's touch keys.

Busy Forward

End of Dial Digit

Set enable or disable to route the call to preset number when the line has no answer or currently online.

Set end of dial key as

, , or None

5.4. PPPoE Configuration

Device

User Name

Password

IP Address

Destination

DNS primary

Reboot After Remote Host

Enable or Disable the PPPoE connection

Enter your PPPoE account

Enter your PPPoE account password

It shows the Internet connection IP address if the gateway PPPoE connection established.

It shows the Internet connection gateway address if the gateway PPPoE connection established.

It shows the Internet connection Domain Name Server IP address if the gateway PPPoE connection established.

The gateway will reboot by self when lost the Internet connection and regain the connection

Disconnection

5.5. Voice Configuration

Codec Priority

Frame Size

G. 723 Silence Suppression

Volume

Set the Codecs priority here. If you set the g723 at first priority, g729a at second priority, then the gateway will use g723 to negotiate the connection first, then shift to second codec if the first didn’t match

Set Specify sending packet size, G.723: 30/60/90, G.711A, G.711U, G.729: 20/40/60/80ms, G.729A: 20/40/60/80ms. The smaller the packet size, the shorter the delay time. If network is in good condition, smaller sending packet size is recommended

Silence Suppression, also called “Voice activation detection” (VAD) is a software application that allows a data network carrying voice traffic over the Internet to detect the absence of audio and conserve bandwidth by preventing the transmission of "silent packets" over the network.

Adjust the volume levels

Echo Cancelor

Voice (Incoming) : 0 ~ 63 Input gain (Outgoing) : 0 ~ 38 DTMF (Keypad tone) : 0 ~ 31

Echo Canceller is designed to cancel acoustic feedback between a loudspeaker and a microphone in loud speaking

audio systems.

Jitter Buffer

It’s a hardware device or software process that eliminates jitter caused by transmission delays in an Internet telephony (VoIP) network. As the jitter buffer receives voice packets, it adds small amounts of delay to the packets so that all of the packets appear to have been received without delays. Voice signals are sequential by nature (i.e., they must be played back in the order in which they were sent) and the jitter buffer ensures that the received packets are in the correct order. Without a jitter buffer to smooth the transmission, data can be lost, resulting in choppy audio signals.

5.6. Phone Pattern Configuration

For tone simulation, FXS Gateway adopts dual frequencies as traditional telephone does. Default tone value is set according to U.S. tone specification. Users may adjust the values to their own country’s tone specification or users-defined tone specification.

Ring Tone

Ring Back Tone

Busy Tone

Dial Tone

2nd Dial Tone

Set Ring frequency, on time, off time. Gateway will give ring to phone set to trigger ring. If user found that phone set cannot ring when having incoming call, please try to increase ring frequency here.

y ringing frequency: 15 ~ 100 (Unit: Hz) y ringing ring ON/OFF: 0 ~ 8000 (Unit: ms) y ringing level: 0 ~ 94 (Unit: V) y tone frequency: 0 ~ 65535 (Unit: Hz) y tone freqLevel: 0 ~ 65535 (Unit: mVrms) y tone Tone ON/OFF: 0 ~ 8000 (Unit: ms)

Set ring back tone parameters

Set busy tone parameters

Set Dial tone parameters

To configure the value of the local 2

dial tone

Audible tones are used in the telephone system to indicate the progress or disposition of a call. Precise dial tone consists of Current day "precise" tones consist of a summation of two low distortion sine waves.

The Dial Tone signal is used in Public Switched Telephone Networks to indicate that the telephone network switching equipment has recognized that a telephone has gone off-hook, and the switching equipment is prepared to receive the dialed digits or DTMF codes.

The Ring-back signal is used in Public Switched Telephone Networks to indicate to the caller that the called number is not busy, and that the line is being "rung" or signaled that an incoming call is present. In most cases, the ring-back signal has the same cadence as the ring generators used in that country, but the ring-back and ring generators are usually not synchronized with one another.

The Busy signal is used in Public Switched Telephone Networks to indicate that the called party is already taking another call. On most switching systems, the busy signal will be emitted until the caller goes on-hook.

Note:

If disconnect tone is single-frequency, user has to configure the same frequency value of “Low frequency” and “High frequency”; the same level of “Low frequency” and “High frequency” For On/Off cadence, user must set “1023” instead of “0”, if there is only one set of cycle, please as in second set columns

5.7. Support Configuration

T.38 FAX

T.38 is an ITU standard for sending FAX across IP networks in a real-time mode. FAX messages are sent as UDP or TCP/IP packets. Enable the T.38 FAX function, the gateway can send or receive the facsimiles. It must enable on both sides, the caller and called party.

5.8. Phone Book Configuration

Phone Book function allows users to define their own numbers, which mapping to real IP address. It is effective only in peer-to-peer mode.

Add Data

Delete Date

Note:

The e164 number defined in phone book will fully carry to destination. It is not just a representative number for destination’s IP Address. In other words, user dial this e164 number to reach destination, destination will receive the number and find out if it is matched to its e164, including Line number in some particular device.

You can record 20 sets of phone book. Enter the Index, Name, IP Address and E.164 No then click the

the new phone book record.

Enter the Index and click phone book record.

button to erase the

to create

5.9. Prefix Configuration

The Prefix function is using the drop and inserts digits

Add Data

Delete Date

Enter the Index, Prefix, Drop enable/disable and Insert then

click the

Enter the Index and click Prefix record.

to create the new Prefix record.

button to erase the

5.10. DSCP Configuration

Assured Forwarding (AF) PHB

Expedited Forwarding (EF) PHB

Default

User Assign Special DSCP Code

Assured Forwarding (AF): Has four classes and three drop-precedence within each class (so a total of twelve codepoints). Excess AF traffic is not delivered with as high probability as the traffic "within profile," which means it may be

demoted but not necessarily dropped DiffServ AF

Expedited Forwarding (EF): Has a single codepoint (DiffServ value). EF minimizes delay and jitter and provides the highest level of aggregate quality of service. Any traffic that exceeds the traffic profile (which is defined by local policy) is discarded

DiffServ EF

Select TOS value as 0.

User can set other unspecified value here

Differentiated Services (DiffServ, or DS) is a protocol for specifying and controlling network traffic by class so that certain types of traffic get precedence - for example, voice traffic, which requires a relatively uninterrupted flow of data, might get

precedence over other kinds of traffic.

By using DiffServ, traffic is classified based on priority. Then the traffic is forwarded using one of three IETF-defined per-hop behavior (PHB) mechanisms. This approach allows traffic with similar service characteristics to be passed with similar traffic guarantees across multiple networks, even if the multiple networks don't provide the same service the same way. This is an important feature because the Internet is really a network of multiple service provider networks.

DiffServ replaces the first bits in the ToS byte with a differentiated services code point (DSCP). The DSCP is then mapped to the PHB. This technique allows service providers to control how the DSCP codepoints are mapped to PHBs, and each time a packet enters a network domain it may be re-marked.

5.11. Password Configuration

Current Password

New Password

Confirm New Password

There is no password as default setting, it is strongly recommended that you change the factory default password of the gateway. All users who try to access the Gateway’s Web-based setup menu will be prompted for the Gateway’s Password. The new Password must not exceed 12 characters in length and must not include any spaces. Enter the new Password a second time to confirm it.

Select root or administrator

Enter the existing password here

Enter the new password

Enter the new password again

5.12. ROM Upgrade

The web configuration provides Update FXS Gateway ROM Version.

FTP/TFTP server IP address

Target File Name

Method

FTP Login Name

FTP Login Password

Target File Type

Note:

Enter the FTP or TFTP Server IP Address

Enter the new firmware’s file name here

Select download method as FTP or TFTP

Enter the FTP Login name (max 14 byte)

Enter the FTP Login password (max 14 byte)

Select download Target File Type on 2M Boot Image, DSP Application Image, DSP Core Image, DSP Test Image different options from the drop-down list box

To upgrade the firmware version, use the Application ROM only in most cases. 2M ROM includes BOOT and APP images.

5.13. Flash Clean

To reset the gateway settings back to factory default

Note: User whose login name is root only executes it. All configurations in [Network Interface] will be kept.

5.14. Commit Data

To save change after configuring FXS Gateway.

5.15. Reboot System

Reboot the FXS Gateway

6. Command List

This section introduces the command line interface and lists all of the commands. You can use the commands to configure the gateway by telnet.

Command Description

help help/man/? [command]

quit quit/exit/close the telnet connection

debug Show debug message

reboot Re-start the gateway

commit Save the change

ifaddr Network address manipulation

time Show current time

ping Connection test command

pbook Phonebook information and configuration

pppoe PPPoE stack manipulation

flash Clean configuration from flash rom

sysconf System information manipulation

sip SIP information manipulation

security Security information manipulation

voice Voice information manipulation

support Special Voice function support manipulation

tos IP Packet ToS/DSCP values

phone Setup of call progress tones and ringing (SLIC control)

bureau Configure the Hotline mode destination

rom Firmware information and update

passwd Password setting information and configuration

prefix Prefix drop/insert information manipulation

6.1. [help]

Type [help], [man] or [?] to show the command list as the table below.

usr/config$ ?

help help/man/? [command] quit quit/exit/close debug show debug message reboot reboot local machine commit commit flash rom data ifaddr internet address manipulation time show current time ping test that a remote host is reachable pbook Phonebook information and configuration pppoe PPPoE stack manipulation flash clean configuration from flash rom sysconf System information manipulation sip SIP information manipulation security Security information manipulation voice Voice information manipulation support Special Voice function support manipulation tos IP Packet ToS/DSCP values phone Setup of call progress tones and ringing (SLIC control) bureau Bureau line information manipulation rom ROM file update passwd Password setting information and configuration prefix Prefix drop/insert information manipulation

6.2. [quit]

Type [quit] will quit and disconnect the Gateway configuration mode.

6.3. [debug]

Open debug message will show up specific information while Gateway is in operation. After executing the debug command, it should execute command [debug -open] as well.

usr/config$ debug

Debug message information and configuration Usage: debug [-add type1 [[type2]...]] | -open | -close | -status

-status Display the enabled debug flags.

-add Add debug flag.

-delete Remove specified debug flag.

-open Start to show debug messages.

-close Stop showing debug messages. Example: debug -add sip msg debug –open

Parameter Usage:

-status

-add

-sip

-msg

-delete

-open

-close

Display the enabled debug flags. Add debug flag SIP related information voice related information Remove specified debug flag Start to show debug messages Stop showing debug messages

For example, user open debug flags including sip, vp, msg.

usr/config$ debug -add sip msg usr/config$ debug -open

usr/config$ debug -status

Current debug type enabled : Debug Mode is open DEBUG-> SIP MSG

6.4. [reboot]

usr/config$ reboot

 Rebooting...It will take 40 seconds....

After [commit] command, type [reboot] to re-start the gateway to take new configurations effective

6.5. [commit]

usr/config$ commit

This may take a few seconds, please wait....

Commit to flash memory ok!

Save changes after configuring Gateway.

6.6. [ifaddr]

Configure and display Gateway network information.

usr/config$ ifaddr

LAN information and configuration Usage: ifaddr [-print]|[-dhcp used]|[-sntp mode [server]] ifaddr [-ip ipaddress] [-mask subnetmask] [-gate defaultgateway] ifaddr [-dns index [dns server address]] [-ipsharing used[ip address]]

-print Display LAN information and configuration.

-ip Specify WAN ip address.

-lanip Specify LAN ip address.

-mask Set Internet subnet mask.

-gate Specify default gateway ip address

-nat Set NAT service flag (On/Off).

-dhcp Set DHCP client service flag (On/Off).

-sntp Set SNTP server mode and specify IP address.

-dns specify IP address of DNS Server.

-timezone Set local timezone.

-ipsharing Specify usage of an IP sharing device and specify IP address.

-server specify EMS Server IP address

-id specify EMS Server ID

-pwd specify EMS Server password

-emstime specify EMS cycle time Note: Range of ip address setting (0.0.0.0 ~ 255.255.255.255). DHCP client setting value (On=1, Off=0). If DHCP set to 'On', Obtain a set of Internet configuration from DHCP server assigned. SNTP mode (0=no update, 1=specify server IP, 2=broadcast mode). Example: ifaddr -ip 210.59.163.202 -mask 255.255.255.0 -gate 210.59.163.254 ifaddr -nat 1 ifaddr -dhcp 1 ifaddr -sntp 1 210.59.163.254 ifaddr -ipsharing 1 210.59.163.254 ifaddr -dns 1 168.95.1.1

Parameter Usage:

-print

-ip

-lanip

Print current IP setting and status Assign the VoIP gateway’s IP address Specify LAN port IP address (For NAT function), use this command setup lanip address assigned to PC or other machine.

Setting IP address provide PC setup Default Gateway Address

-mask

-gate

-nat

Assign the VoIP gateway’s Subnet Mask Assign the VoIP gateway’s default gateway Provides Network Address Translation function

Enable the NAT function when share the connection to your PCs.

-dhcp

-sntp

Dynamic host configuration (0=Off, 1=On) Simple Network Time Protocol (0=No update, 1=Specify server IP). When SNTP function is activated, users have to specify a SNTP server as network time source

Example : ifaddr -sntp 1 168.95.192.12

-dns

Specify the DNS server’s IP address

-timezone

-ipsharing

Set local time zone according to GMT Enable this function when the VoIP gateway behind the NAT router or IP Sharing devices.

Example : ifaddr -ipsharing 1 61.219.198.204

Note : If you don’t have static public IP address, then the dedicated IP address is not necessary in the command, for example : ifaddr -ipsharing 1 However, dynamic IP

Address is not working in Peer-to-Peer mode.

-server

-id

-pwd

-emstime

specify EMS Server IP address specify EMS Server ID specify EMS Server password specify EMS cycle time

The EMS (Element Management System) is expressly built to simplify deployment, configuration and management of network equipment and to help you streamline delivery of the high-demand services and capabilities enabled.

Note:

One Group only use only LAN IP address, if have two gateway on this group, you must change second gateway LAN IP Address different first gateway.

Gateway First: usr/config$ ifaddr -lanip 192.168.124.124

Gateway Second: usr/config$ ifaddr -lanip 192.168.124.125

Information Example:

usr/config$ ifaddr -print

Internet address information WAN IP address : 192.168.0.243 Subnet mask : 255.255.255.0 Default gateway : 192.168.0.1 NAT enabled : OFF DHCP startup : OFF SNTP : mode=1 server 168.95.195.12 time zone : GMT+8 cycle=1024 mins

IPSharing : no IPSharing device.

Primary DNS Server : 168.95.1.1 Secondary DNS Server : 168.95.1.1

EMS IP Address : 192.168.1.1 EMS User ID : vwusr EMS Password : vwusr EMS cycle time : 0

6.7. [time]

When SNTP function of Gateway is enabled and SNTP server can be found as well, type [time] command to show current network time.

usr/config$ time Current time is WED DEC 01 12:38:38 2004

6.8. [ping]

ping is the name of a computer network tool used on TCP/IP networks (such as the

Internet). It provides a basic test of whether a particular host is operating properly and is reachable on the network from the testing host. It works by sending ICMP packets to the target host and listening for replies For example: if 192.168.1.2 is not existing while 192.168.123.100 exists. Users will have the following results:

usr/config$ ping 192.168.1.2

PING 192.168.1.2: 56 data bytes no answer from 192.168.1.2

usr/config$ ping 192.168.123.100

PING 192.168.123.100: 56 data bytes 64 bytes from 192.168.123.100: icmp_seq=0. time=5. ms 64 bytes from 192.168.123.100: icmp_seq=1. time=0. ms 64 bytes from 192.168.123.100: icmp_seq=2. time=0. ms 64 bytes from 192.168.123.100: icmp_seq=3. time=0. ms

----192.168.123.100 PING Statistics---4 packets transmitted, 4 packets received, 0% packet loss round-trip (ms) min/avg/max = 0/1/5

6.9. [pbook]

Phone Book function allows users to define their own numbers, which mapping to real

IP address. It is effective only in peer-to-peer mode. When adding a record to Phone

Book, users do not have to reboot the machine, and the record will be effective immediately.

usr/config$ pbook

Phonebook information and configuration Usage: pbook [-print [start_record] [end_record]] pbook [-add [ip ipaddress] [name Alias] [e164 phonenumber]] pbook [-search [ip ipaddress] [name Alias] [e164 phonenumber]] pbook [-insert [index] [ip ipaddress] [name Alias] [e164 phonenumber] [port numb er]] pbook [-delete index] pbook [-modify [index] [ip ipaddress] [name Alias] [e164 phonenumber] [port numb er]]

-print Display phonebook data.

-add Add an record to phonebook.

-search Search an record in phonebook.

-delete Delete an record from phonebook.

-insert Insert an record to phonebook in specified position.

-modify Modify an exist record. Note: If parameter 'end_record' is omited, only record 'start_record' will be disp lay. If both parameters 'end_record' and 'start_record' are omited, all records will be display. Range of ip address setting (0.0.0.0 ~ 255.255.255.255). Range of index setting value (1~100), Example: pbook -print 1 10 pbook -print 1 pbook -print pbook -add name Test ip 210.59.163.202 e164 1001 pbook -insert 3 name Test ip 210.59.163.202 e164 1001 pbook -delete 3 pbook -search ip 192.168.4.99 pbook -modify 3 name Test ip 210.59.163.202 e164 1001

Parameter Usage:

-print

Print out current contents of Phone Book. Users can also add index number, from 1 to 50, to the parameter to show specific phone number.

-add

add a new record to phone book. When adding a record, users have to specify name, IP, and e164 number to complete the command.

name e164 ip port drop

Name to represent caller. E.164 number for mapping with IP address of caller IP address of caller Call signal port number of caller Drop e.164 number when dial out. 0 means to keep e.164 number, 1 means to drop e.164 number when dialing out.

insert

-modify

Insert digits.(1~10 digits)

modify an existing record. When using this command, users have to specify the record’s index number, and then make the change.

-delete

delete a specific record. For example : pbook -delete 3

Note:

Index number: means the sequence number in phone book. If users do request a specific index number in phone book, Gateway will give each record a automatic sequence number as index.

PhoneBook Rules:

For example:

usr/config$ pbook -print

index Name IP E164 Port ====================================================================== 1 SP5100 192.168.0.242 5100

----------------------------------------------------------------------

6.10. [pppoe]

Display PPPoE related information.

usr/config$ pppoe

PPPoE device information and configuration Usage: pppoe [-print]|[-open]|[-close] pppoe [-dev on/off][-id username][-pwd password][-reboot on/off]

-print Display PPPoE device information.

-dev Enable(=1) or Disable(=0) device.

-open Open PPPoE connection.

-close Disconnect PPPoE connection.

-id Connection user name.

-pwd Connection password.

-reboot Reboot after remote host disconnection.

Parameter Usage:

-print

-dev

-open

-close

-id

-pwd

-reboot

print PPPoE status. Enable or Disable PPPoE Dial-up function Open the connection Disconnect the connection The User name ID provided by ISP The Login password provided by ISP Reboot the gateway after the PPPoE connection disconnected

6.11. [flash]

Restore the gateway’s configurations back to default.

usr/config$ flash

Flash memory information and configuration Usage: flash -clean

Note: This command will clean the configuration stored in the flash and reboot it.

Parameter Usage:

-clean

clean all the user defined value, and reboot Gateway in factory default mode

Note:

It is recommended that use [flash -clean] after application firmware upgraded. User whose login name is root only executes it. All configurations in command [ifaddr] and [pppoe] will be kept.

6.12. [sysconf]

This command displays system information and configurations.

usr/config$ sysconf

System information and configuration Usage: sysconf [-print] [-idtime digit] [-bf digit] [-keypad dtmf] [-faxtype type][-2833type type][-lcdrop ON/OFF] [-droptime digit][-eod digit] [-callerid type] [-service used][-dtmfstart digits] [-dtmfend digits] sysconf -print

-print Display system overall information and configuration.

-idtime Inter-Digits time.(1~10 sec)

-service Specify gateway service type. (0: Dial in service, 1: HotLine service.)

-keypad Select DTMF type: 0=In-band, 1=RFC2833.

-faxtype FAX Payload Type (range:96~128 inter-used:100,102~105)

-2833type RFC2833 Payload Type (range:96~128 inter-used:100,102~105)

-lcdrop Disconnect Supervision(Loop Current Drop) (ON:1 / OFF:0)

-droptime Period of Loop Current Drop (ms)

-eod End of Dial Digit setting(0: none, 1: *, 2: #)

-callerid Caller ID Type setting, 0: Disable, 1: FSK(BELLCORE), 2: DTMF, 3: NTT.

-dtmfstart DTMF CallerID Start Symbol.

-dtmfend DTMF CallerID End Symbol. Example: sysconf -keypad 0 -eod 2 -callerid 1

Parameter Usage:

-print

-idtime

-service

-keypad

-faxtype

-2833type

-lcdrop

Show the sysconf current status. Set the duration (in second) of two pressed digits in dial mode as timed out. If after the duration user hasn’t pressed next number, it will dial out all number pressed (1-10 seconds). Specify gateway service type. (0: Dial in service, 1: HotLine service.) DTMF replay type. When value is “1”, FXS Gateway will transfer DTMF signal via RTP payload as defined in RFC2833. When the value is set to “0”, the DTMF type is set as In-band. FAX Payload Type. Rrange:96~128 inter-used:100,102~105. RFC2833 Payload Type. Range: 96~128 inter-used: 100, 102~105. Disconnect Supervision (Loop Current Drop) (ON:1 / OFF:0).

-droptime

Period of Loop Current Drop (ms).

-eod

-callerid

Select the End of Dial key, “#”, “*” or none Select the Caller ID type, 0 = disable, 1 = FSK(Bell core), 2 = DTMF, 3 = NTT. After the first ring at destination site, device will send line number as caller ID to called site.

-dtmfstart

-dtmfend

DTMF Caller ID Start Symbol DTMF Caller ID End Symbol

Payload Type, the essential data that is being carried within a packet or other transmission unit. The payload does not include the "overhead" data required to get the packet to its destination. Note that what constitutes the payload may depend on the point-of-view. To a communications layer that needs some of the overhead data to do its job, the payload is sometimes considered to include the part of the overhead data that this layer handles. However, in more general usage, the payload is the bits that get delivered to the end user at the destination.

6.13. [sip]

This command is to configure SIP related parameters.

usr/config$ sip

SIP stack information and configuration Usage: sip [-print] [-mode pxmode] [-outpx IPaddmress][-transport type] sip [-px address] [-px2 address] [-pxport number] [-prefix prefixstring] [-line1 number] [-expire t1] [-port udpPort] [-rtp rtpPort] sip -print

-print Display SIP stack information and configuration.

-mode Configure as Peer-to-Peer mode:0/Proxy mode:1.

-px Primary Proxy server address. (IPv4 address or dns name)

-px2 Secondary Proxy server address. (IPv4 address or dns name)

-pxport Proxy server port. (the port of proxy)

-outpx OutBound Proxy server address. (IPv4 address or dns name)

-prefix Specify prefix string, use it when UserID contains alphabets (if UserID uses numerals, specify as null)

-line1 TEL1 Phone number.

-pbsearch Search phone book 0:off/1:on.

-expire The relative time after which the message expires(0 ~ (2^31-1))

-port SIP local UDP port number (5060~5070), Default: 5060

-rtp RTP port number (2326~65534), Default: 16384 Example: sip -mode 1 sip -px 210.59.163.171 -line1 70

Parameter Usage:

-print

-mode

-px

-px2

-pxport

-outpx

-prefix

-line1

-pbsearch

Show the SIP current settings Select the P2P mode or Proxy mode, 0 = P2P, 1 = Proxy To specify Proxy address when FXS Gateway is in proxy mode. Proxy address can be IPv4 address or DNS name. To specify Secondary Proxy server address. To configure proxy server signaling port, default value is 5060, if there is no special request of Proxy server, please don’t change this value. Set IP Address or URL address (Domain Name Server must be configured. Please refer to Network Configure) of outbound Proxy server. when your username contains alphabets, for example sip1123, then specify the prefix string as “sip”. Assign gateway’s line number enable/disable phone book search function under Proxy Mode. If

user enabled this function, the gateway will search dialed number in phone book to see if there is any matched table before send to Proxy server, and if there is a matched data in phone book, the gateway will make call to related IP address.

-expire

This parameter set duration time for sending registration information.

-port

-rtp

SIP port which used to listen incoming SIP messages Specify the RTP received port number

6.14. [security]

This command is used to configure the account information included username and password obtained from the proxy service provider

usr/config$ security

Security information and configuration Usage: security [-line number][-name username] [-pwd password] security [-print]

-print Display system account information and configuration.

-line Specify which line number you want to set the account.

-name Specify user name.

-pwd Specify password. Example: security -line 1 -name 1001 -pwd 1001

Parameter Usage:

-print

-line

-name

-pwd

Shows the current settings Specify the line for the account configuration, here has only one line for this gateway model. Specify the username of your account information. Specify the password of your account information.

6.15. [voice]

The voice command is associated with the audio setting information.

usr/config$ voice

Voice codec setting information and configuration Usage: voice [-send [G723 ms] [G711U ms] [G711A ms] [G729 ms] ] [-volume [voice level] [input level] [dtmf level]] [-nscng [G711U used1] [G711A used2] [G723 used3]] [-echo used] [-mindelay t1] [-maxdelay t2] voice -print voice -priority [G723] [G711U] [G711A] [G729]

-print Display voice codec information and configuration.

-send Specify sending packet size. G.723 (30/60 ms) G.711U (20/40/60 ms) G.711A (20/40/60 ms) G.729 (20/40/60/80 ms)

-priority Priority preference of installed codecs. G.723 G.711U G.711A G.729

-volume Specify the following levels: voice volume (0~63, default: 25), input gain (0~38, default: 25), dtmf volume (0~31, default: 23),

-nscng No sound compression and CNG. (G.723.1 only, On=1, Off=0).

-echo Setting of echo canceller. (On=1, Off=0, per port basis).

-mindelay Setting of jitter buffer min delay. (0~150, default: 90).

-maxdelay Setting of jitter buffer max delay. (0~150, default: 150). Example: voice -send g723 60 g711u 60 g711a 60 g729 60 voice -volume voice 20 input 32 dtmf 27 voice -echo 1

Parameter Usage:

-print

-send

Shows the current settings

To define packet size for each codec. 20/40/60/80 ms means to send a voice packet per 20/40/60/80 milliseconds. The smaller the packet size, the shorter the delay time. If network is in good condition, smaller sending packet size is recommended. In this parameter, 20/40/60ms is applicable to G.711u/a law, 20/40/60ms is applicable to G.729 codec, while 30/60ms is applicable to G.723.1 codec.

-priority

Codec priority while negotiating with other SIP device. The codec listed in left side has the highest priority when both parties determining final codec. For example : usr/config$ voice -priority g729 g723 g711u g711a (Selected four Codecs, G.729 is the first choice)

-volume voice input dtmf

-nscng

-echo

-mindelay

-maxdelay

Note:

To adjust the voice, input and dtmf levels which can be heard from Gateway side(range 0~63, default: 25). which the opposite party hears (range 0~38, default: 25). which sends to its own Line (range 0~31, default: 23).

Silence suppression and comfort noise generation setting (1 = ON; 0 = OFF). It is applicable to G.723 codec only.

Enable or Disable the echo cancellation

The minimum jitter buffer size (Default value= 90 ms).

The minimum jitter buffer size (Default value= 150 ms).

Be sure to know well the application before you change voice parameters because this might cause incompatibility.

6.16. [support]

This command provides some extra functions that might be needed by users.

usr/config$ support

Special Voice function support manipulation Usage: support [-t38 enable] [-busy number] [-noanswer number] [-uncon number] support -print

-t38 T.38(FAX) enabled/disabled.

-busy Busy Forward number.(if empty, please fill "null")

-noanswer No Answer Forward number.(if empty, please fill "null")

-uncon Unconditional Forward number.(if empty, please fill "null")

Example: support -t38 1 support -busy 1001 support -uncon null

Parameter Usage:

-print

-t38

-busy

-noanswer

-uncon

Shows the current settings Enable or disable FAX ability. The function is will automatically defer codec (G.723 or G.729a) to T.38 when FAX signal is detected.

Provide setting busy forward to other number, when your gateway is setting this function, it will forward to setting phone number if the channel is busy,

Provide setting noanswer forward to other number, when you set this function, it will forward to setting phone number if no one answer the call.

Provide setting Unconditional forward to other number, when you set this function, all the calls to your number will forward to setting phone number.

6.17. [tos]

IP Packet ToS (Type of Service)/ Differentiated Service configuration.

usr/config$ tos

IP Packet ToS(type of Service)/Differentiated Service configuration Usage: tos [-rtptype dscp] tos [-sigtype dscp] tos -print [-rtpreliab mode] tos -print

Example: tos -rtptype 7 -sigtype 0

-rtptype

-sigtype

the packages of voice (0~63) the package of call signal (0~63)

IPv4 Head Format

In RFC 791, the following 8 bits were allocated to a Type of Service (ToS) field - now DiffServ and ECN. For instance, one host could set its IPv4 datagrams' ToS field value to prefer low delay, while another might prefer high reliability. In practice, the ToS field has not been widely implemented. However, a great deal of experimental, research and deployment work has focused on how to make use of these eight bits. These bits have been redefined and most recently through DiffServ working group in the IETF and the Explicit Congestion Notification codepoints

Note:

The value of rtptype and sigtype is from 0 to 63. ToS only works if it has related network devices supported.

6.18. [phone]

Gateway progress tone is configurable. Default tone value is set according to U.S. tone specification. Users may adjust the values according to their own country’s tone specification or users-defined tone specification.

usr/config$ phone

Phone ringing , ringback tone , busy tone , dial tone setting and notes Usage:

phone [-ring [freq ] [ringON ] [ringOFF ] [ringLevel]] [-rbt [freqHi ] [freqLo ] [freqHiLev] [freqLoLev] [Tone1ON] [Tone1OFF] [Tone2ON ] [Tone2OFF ]] [-bt [freqHi ] [freqLo ] [freqHiLev] [freqLoLev] [Tone1ON] [Tone1OFF] [Tone2ON ] [Tone2OFF ]] [-dt [freqHi ] [freqLo ] [freqHiLev] [freqLoLev] [Tone1ON] [Tone1OFF] [Tone2ON ] [Tone2OFF ]] [-flash [freqLo ] [freqHi ]] [-level [loopCurrentLevel] [onhookLineVoltageLevel ]] phone [-print [ring]|[rbt]|[bt]|[dt]|[flash]]

-print Display phone ringing/tone configuration.

ring : ringing rbt : ringback tone bt : busy tone dt : dial tone flash: flash tone

-ring ringing configuration set .

-rbt ringback tone configuration set .

-bt busy tone configuration set .

-dt dial tone configuration set .

-flash flash configuration set .

-level Loop Current and On-Hook Line Voltage level set .

Note: ringing frequency : 15 ~ 100 (Unit : Hz) ringing ring ON/OFF : 0 ~ 8000 (Unit : ms) ringing level : 0 ~ 94 (Unit : V) tone frequency : 0 ~ 65535 (Unit : Hz) tone freqLevel : 0 ~ 65535 (Unit : mVrms) tone Tone ON/OFF : 0 ~ 8000 (Unit : ms) level loopCurrent : 0 ~ 7 (20mA ~ 41mA, Step : 3mA) level OnHookVol : 0 ~ 63 ( 0V ~ 94.5V, Step : 1.5V) Example: phone -print rbt phone -ring 20 2000 4000 94 phone -rbt 480 440 125 105 2000 4000 2000 4000 phone -bt 620 480 125 105 500 500 500 500 phone -dt 440 350 96 96 8000 0 8000 0 phone -flash 400 800 phone -level 1 32

Parameter Usage:

-print

-ring

-rbt

-bt

Specify which tone settings you want to display

ring : ring tone settings rbt : ring back tone settings bt : busy tone settings dt : dial tone settings flash : flash time settings

To set RING tone value. The played tone type, when Gateway is receiving a call.

To set Ring Back Tone value. The played tone type, when Gateway receives a Q.931 Alerting message. In condition that Gateway is the originate side.

To set Busy Tone value. The played tone type, when destination is busy.

-dt

To set Dial Tone value. The played tone type, when hook off a phone set of workable Gateway.

-flash

Set the detective flash range in ms, for example, 400-800 ms.

-level

Loop Current and On-Hook Line Voltage level set.

Note:

For tone simulation, Gateway adopts dual frequencies as traditional telephone does. If users want to have their own call progress tone, they can change the value of tones. High and Low frequency/level/cadence can be configured respectively.

6.19. [bureau]

To set Hotline function must be under Peer-to-Peer mode and switch to hotline mode.

usr/config$ bureau

Bureau line setting information and configuration Usage: bureau [-hotline [Port DestIP TELnum]] bureau -print

-print Display Bureau line information and configuration.

-hotline Set Hot line information. (Port range: 1~6)

Note: Hotline feature should be used together with: $sysconf -service 1 (HotLine service) $sip -mode 0 (peer-to-peer mode) Example: bureau -hotline 1 192.168.4.69 628

Parameter Usage:

-print

Shows the current settings

-hotline

Define Line Hotline table respectively. The table is included [Line number], [destination IP Address] and [destination Port or Number].

For example

1. Destination is a FXS device, 628 is its Line1 number

usr/config$ bureau -hotline 1 200.168.4.69 628

User picks up the telephone handset connects to gateway, and then hears the ringback tone generated from destination. Of course, the destination line 628 is ringing simultaneously.

2. Destination is a FXO device, Port_1 has connected to PSTN Line.

usr/config$ bureau -hotline 1 200.168.4.69 82265699

User picks up the Line1, and then hears the ringback tone generated from destination. Simultaneously, 82265699 numbers is the destination, which is dialed from Port_1 (Above FXO example is subject to the FXO configurations, such as 2nd dial ON or OFF.)

6.20. [rom]

ROM file information and firmware upgrade function.

usr/config$ rom

ROM files updating commands Usage: rom [-print] [-app] [-boot] [-dsptest] [-dspcore] [-dspapp] [-ht] [-method used] [-boot2m]

-s TFTP/FTP server ip -f filename

rom -print

-print show versions of rom files. (optional)

-app update main application code(optional)

-boot update main boot code(optional)

-boot2m update 2M code(optional)

-ht updata Hold Tone PCM file(optional)

-dsptest update DSP testing code(optional)

-dspcore update DSP kernel code(optional)

-dspapp update DSP application code(optional)

-s IP address of TFTP/FTP server (mandatory)

-f file name(mandatory)

-method download via TFTP/FTP (TFTP: mode=0, FTP: mode=1)

-ftp specify username and password for FTP

Note: This command can run select one option in 'app', 'boot', , 'dsptest', 'dspcore', and 'dspapp'. Example: rom -method 1 rom -ftp vwusr vwusr rom -app -s 192.168.4.101 -f app.bin

Parameter Usage:

-print

-app

-boot

-boot2m

-ht

-dsptest

-dspcore

-dspapp

-s

-f

-method

-ftp

Shows the current settings update application program code update boot code Includes APP and Boot code updata Hold Tone PCM file(optional) update DSP testing code(optional) update DSP kernel code(optional) update DSP application code(optional) To specify TFTP / FTP server IP address for upgrading To specify the target file name, this will replace the old one. To decide using TFTP or FTP as file transfer server. TFTP = 0 , FTP = 1 If users choose FTP in above item, it is necessary to specify pre-defined username and password when upgrading files.

6.21. [passwd]

For security concern, users have to input the password before entering configuration mode. [passwd] command is for password setting purpose.

usr/config$ passwd

Password setting information and configuration Usage: passwd -set Loginname Password passwd -clean Note:

1. Loginname can be only 'root' or 'administrator'

2. passwd -clean will clear all passwd stored in flash,

please use it with care. Example: passwd -set root Your_Passwd_Setting

Parameter Usage:

-set

Set login name and password, input login name then input new password.

-clean

Clear all password setup, and change null.

Note:

Gateway Login name only use root or administrator. Both accounts have the same authorization, except commands that can be executed by login name root only

[passwd -set root], [rom -boot], [room -boot2m] and [flash -clean].

6.22. [prefix]

Prefix drop/insert information manipulation

Prefix drop/insert information and configuration Usage: prefix -add [prefix number][drop number][insert digits] prefix -delete index prefix -modify index [prefix number][drop number][insert number] prefix -print Prefix drop/insert information. prefix The prefix of dialed number. drop Drop prefix(Enable:1/Disable:0). insert Insert digits. Example: prefix -add prefix 100 drop 1 insert 2000 prefix -add prefix 100 drop 1 prefix -add prefix 100 drop 0 insert 200 prefix -delete 1 prefix -modify 1 prefix 100 drop 0 insert 300

Parameter Usage:

-add

-modify

-delete

Add a rule to drop or insert prefix digits of incoming call.

prefix : Set which prefix number to implement prefix rule. drop : Enable or disable drop function. If this function is enabled,

Gateway will drop prefix number on incoming call.

insert : Set which digit to insert on incoming call.

Modify a rule to drop or insert prefix digits of incoming call.

Delete a rule to drop or insert prefix digits of incoming call.

MicroNet Technology SP5001A-S User Manual

Specifications and Main Features

Frequently Asked Questions

User Manual

2. Setting Up the Gateway

2.1. Connecting the SP5001A/S

2.2. Internet Connection Setup

2.3. Proxy Mode Setup

2.4. Peer-to-Peer Mode Setup

2.5. P2P Connection Example

3.1. Behinds the NAT Router (P2P Mode)

Setup the SP5001A/S:

Setup the NAT router:

A. DMZ Setup

B. Port Forwarding (Virtual Server)

4.1. TFTP Server Setup

4.2. Upgrade by WEB Interface

4.3. Upgrade by Telnet Command

5. WEB Configuration Menu

5.6. Phone Pattern Configuration

5.8. Phone Book Configuration