User’s Manual IP Telephony Gateway Model No.: SP5001C, SP5001D, SP5002A, SP5012A World Wide Web: www.micronet.com.tw ; www.micronet.
About this User’s Manual This User’s Manual gives users basic steps on installation and operation. Please read this manual chapter by chapter. Chapter 1. Introduction Introduce the IP Telephony Gateway to users in terms of feature, appearance, and application. Chapter 2. Startup Help user complete basic configuration. Chapter 3. Operation Show user how to use the device to process phone call and FAX. Chapter 4. Web Administration Provide command reference of Web Interface for advanced setting. Chapter 5.
Table of Content 1. ..Introduction.............................................................................................. 4 1.1 Key Features ................................................................................................ 5 1.2 Physical Description ..................................................................................... 6 1.3 Application.................................................................................................. 10 2. ..Startup ................
.4 4.3.7 DMZ * ............................................................................................................ 34 4.3.8 Virtual Server * .............................................................................................. 34 SIP Settings................................................................................................ 35 4.4.1 Service Domain ............................................................................................. 35 4.4.2 Port Settings......
1. Introduction Micronet SP5001C, SP5001D, SP5002A and SP5012A IP Telephony Gateway is designed to connect standard telephone devices to IP-based telephony networks, providing users with high-quality VoIP service. SP5001C / SP5002A provides: ● 1/2 FXS port(s) for phone set, FAX machine, or PBX’s trunk SP5001D provides: ● 1 FXS port for phone set, FAX machine, or PBX’s trunk ● 1 PSTN port for PSTN lifeline that tranceives PSTN calls as backup even if VoIP fails.
1.1 Key Features z z z z z z z z z z z z z z Compliant with IETF SIP standards Provide 2 10/100M RJ-45 ports for WAN and LAN connection Support G.729a/b, G.711a/µ-law, and G.726 codecs Support up to 3 SIP service domains Support STUN and Outbound proxy for NAT traversal Support VAD, CNG, EC, and Adaptive Jitter Buffer Support FSK / DTMF caller ID display Support Call Hold / Call Waiting / Call Forward Support 3-way conference Provide phone address book and speed dialing function Transmit voice and FAX (T.
1.2 Physical Description SP5001C: SP5001C Front Panel LED Status Description PWR On/Green Power On STATUS On/Amber Line Registered TEL On/Amber Phone set off-hook LAN On/Green Link On WAN On/Green Link On SP5001C Rear Panel --------------------------------------------------------------------------------------------------RESET Factory default button.
SP5001D: SP5001D Front Panel LED Status Description PWR On/Green Power On PSTN On/Amber PSTN mode / VoIP Unregistered TEL On/Amber Phone set off-hook LAN On/Green Link On WAN On/Green Link On SP5001D Rear Panel --------------------------------------------------------------------------------------------------RESET Factory default button.
SP5002A: SP5002A Front Panel LED Status Description PWR On/Green Power On TEL1 On/Amber Line Registered TEL2 On/Amber Line Registered LAN On/Green Link On WAN On/Green Link On SP5002A Rear Panel --------------------------------------------------------------------------------------------------RESET Factory default button.
SP5012A: SP5012A Front Panel LED Status Description PWR On/Green Power On TEL On/Amber Line Registered LINE On/Amber Line Registered LAN On/Green Link On WAN On/Green Link On SP5012A Rear Panel --------------------------------------------------------------------------------------------------RESET Factory default button.
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SP5001D: ● 1 RJ-11 FXS port is provided for phone set or PBX's trunk line connection ● 1 RJ-11 PSTN port is provided to tranceive PSTN calls as backup even if VoIP fails ● ● 2 RJ-45 ports of 10/100M are provided for WAN and LAN connection The IP telephony gateway can share Internet access with LAN clients SP5012A: ● 1 RJ-11 FXS port is provided for phone set or PBX's trunk line connection ● 1 RJ-11 FXO port is provided for PSTN or PBX's extension connection, and for PSTN client to communicate wit
VPN (Virtual Private Network) The IP telephony gateway series supports PPTP client for VPN function. It can establish VPN tunnel with PPTP server, and get access to the peer private network as if it is located in the same LAN. For special condition that SIP proxy server is located in the private network of CO site, the gateway can register, and request a phone call via the VPN tunnel. Please refer to the section 2.4 VPN Settings.
2. Startup 2.1 Login into the System The embedded web configuration allows you to use a web browser to manage the IP Telephony gateway. Step 1. Connect LAN port to your managing PC. Or, connect the gateway with PC by hub/switch. Step 2. Launch your web browser with http://192.168.123.1:9999/. Please configure IP address of PC with 192.168.123.x. Step 3. The Password screen now appears. Type “root” in the user name field and your password (none by default) in the password field. Step 4. Click on Login.
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2.2 Network Configuration By default, the gateway is in NAT mode (router mode) and can share Internet access with PCs. Go to [ Network / WAN Settings ], and configure WAN setting according to actual condition. In default IP type of DHCP client, it requests necessary IP information from your ISP automatically. ----------------------------------------------------------------------------------------------------Note: Different ISPs require different methods of connecting to the Internet.
----------------------------------------------------------------------------------------------------Parameter Description ----------------------------------------------------------------------------------------------------LAN Mode Bridge: pure VoIP gateway NAT: VoIP router IP Type Select Fixed IP, DHCP (default), or PPPoE IP IP address provided by ISP Mask Subnet mask provided by ISP Gateway ISP’s IP address gateway DNS Server1/2 IP address of primary/secondary DNS server MAC MAC address PPP
2.3 SIP Configuration Go to [ SIP Settings / Service Domain ]. Each port can be configured to register 3 different service domains. ----------------------------------------------------------------------------------------------------Parameter Description ----------------------------------------------------------------------------------------------------Phone No.
----------------------------------------------------------------------------------------------------Note: Please save and reboot the system to take effect. Go to [ Save Change ] to save configuration, and the system will reboot automatically.
2.4 VPN Configuration IP telephony gateway supports PPTP client for VPN. It can establish VPN tunnel with PPTP server. Go to [ Network / PPTP Settings ], and set PPTP server address, and authentication information (username, password).
After tunnel is established, the gateway gets one private IP address (Interface 2) from PPTP server as shown in [ Network / Status ].
2.5 DDNS DDNS allows you to map the static domain name to a dynamic IP address. You must get an account, password and your static domain name from the DDNS service providers. ● DDNS makes the gateway accessible for other client to call in P2P (peer-to-peer) mode, when the IP address is dynamic. ● DDNS setting is not necessary when IP call is tranceived via service domain only. Go to [ Network / DDNS Settings ] and set up DDNS.
3. Operation 3.1 Make a Call By default, call is sent via the first registrar server only. The telephone number of clients in the second/third service domain will be not accessible. Make a Call Press + # ----------------------------------------------------------------------------------------------------Note: ● Once the first registration fails, the second realm will be activated. ● The gateway can always receive incoming call from the client of either registered service domain.
3.2 Speed Dial / P2P call Speed dial Press + # Go to [ Phone Book / Speed Dial Setting ]. User can create 10 entries (0~9) in Speed Dial Phone List.
3.3 Call Forward Go to [ Phone Setting / Call Forward ]. There are 3 selections in Forward type. User must select the condition under which to forward calls. ----------------------------------------------------------------------------------------------------Parameter Description ----------------------------------------------------------------------------------------------------All Forward Forward the call in any conditions ● Off: call forward disabled. ● On/IP: Call forward to IP.
3.4 Call Hold / Call Waiting / Conference The IP telephony gateway provides telephony features, as call hold, call waiting, and 3-way conference. Call Hold: Hold a existing call Call waiting: Hold a existing call, and answer a new incoming call 3-way conference: Talk with other 2 party in the same session.
3.5 FAX Send FAX ● Press to connect fax machine ● Start to send FAX T.38 FAX: Go to [ Phone Setting / T.38 (FAX) Setting ]. Click on “On” and enable T.38 to tranceive FAX over IP. In-band FAX: disable T.38 and choose G.711 codec as top priority. Please refer to the section 4.4.3 Codec Settings. ----------------------------------------------------------------------------------------------------Note: When sending in-band FAX (in G.711), please disable T.38 and choose G.
4. Web Administration 4.1 Phone Book Please refer to the section 3.2 Speed Dial / P2P call. 4.2 Phone Setting 4.2.1 Call Forward Please refer to the section 3.3 Call Forward. 4.2.2 SNTP Settings User can set up the primary and second SNTP Server IP Address, to get the date/time information. 4.2.3 Volume Settings User can set up the Handset Volume, Ringer Volume, and the Handset Gain. ● ● ● Handset Volume: adjust the volume that you hear from the handset.
4.2.4 Block Settings User can set up the gateway to block incoming calls and the period. 4.2.5 Caller ID User can set the device to show Caller ID in your PSTN Phone. The gateway supports FSK and DTMF. 4.2.6 Dial Plan Setting User can set dialing plan and timeout to send a phone call after dialing number is input.
----------------------------------------------------------------------------------------------------Parameter Description ----------------------------------------------------------------------------------------------------Replace Prefix code On: enable “Replace rule” Off: disable “Replace rule” Replace rule Replace matched prefix with another Dial Plan If dialed numbers match the rule, numbers is sent out. If not, numbers would not be sent out.
* means: keypad＊ on the phone x means: digit 0, 1, 2~9 # means: keypad # on the phone + means: or Auto prefix: 02 (0000~9999) Input Sent out 22183656 02-22183656 82265630 02-82265630 Prefix Unset Plan: 1 + 0 + xxxxxx ● With prefix “0”, auto prefix “02” is not prepended to dialing number ● With prefix “1”, auto prefix “02” is not prepended to dialing number ● With 6-digit number, auto prefix “02” is not prepended to dialing number Auto Prefix Input Sent out 02 0075 0075 02 1075 1075 02
4.2.9 T.38 (FAX) Setting Please refer to the section 3.5 FAX.
4.3 Network 4.3.1 Status User can check the current Network setting. 4.3.2 WAN Settings Please refer to the section 2.2 Network Configuration. 4.3.3 LAN Settings User can configure for LAN clients.
----------------------------------------------------------------------------------------------------Parameter Description ----------------------------------------------------------------------------------------------------IP IP address of LAN port. Default gateway IP of LAN clients in the local network Mask Subnet mask MAC MAC address DHCP Server “On” means DCHP server enabled.
4.3.6 PPTP Settings Please refer to the section 2.4 VPN Configuration. 4.3.7 DMZ * Open another page http://192.168.123.1:9999/dmzset.htm. With the function enabled, the gateway will re-direct all packets going to your WAN port IP address to a particular IP address in your LAN. * will be available in the later version. 4.3.8 Virtual Server * Open another page http://192.168.123.1:9999/vsset.htm.
4.4 SIP Settings 4.4.1 Service Domain Please refer to the section 2.3 SIP Configuration. 4.4.2 Port Settings User can set up SIP and RTP ports. 4.4.3 Codec Settings User can set up the Codec priority, RTP packet length, and VAD function. Please follow service provider’s suggestion to configure.
4.4.4 Codec ID Settings User can set up codec ID for different codec. This ID represents the codec used to encode data in the Track. 4.4.5 DTMF Setting User can set up the method of DTMF transmission: In-band, RFC2833, or SIP Info. 4.4.6 RPort Setting User can set up the RPort Enable/Disable.
Symmetric Response Routing. This behavior is not desirable in many cases. Please configure it according to your service provider. 4.4.7 Other Settings User can set up the Hold by RFC, Voice/SIP QoS and SIP expire time in this page.
EF (Expedited Forwarding). ----------------------------------------------------------------------------------------------------SIP expire time the time used to inform proxy server of the valid duration of registration information. ----------------------------------------------------------------------------------------------------- 4.5 NAT Trans. / STUN Please refer to the section 2.3 SIP Configuration.
4.6 Others 4.6.1 Auto Config User can disable Auto Configuration or enable the function by TFTP/FTP. Please contact with your service provider for necessary information. 4.6.2 ICMP Setting User can set the gateway to reply ICMP echo request or not. Setting this function to “ON”, you will get reply when you PING this gateway. Setting this function to “Off”, you get no reply when you PING this gateway. 4.6.3 PTT Setting Select the PTT setting for FXS interface by different country.
4.7 System Auth. Change system login name and password. 4.8 Save Change Click on the Save button. The system will automatically restart and the new setting will take effect. 4.9 Update 4.9.1 New Firmware User can upgrade the system via TFTP or HTTP in this page.
Click on the Update button to start upgrading. 4.9.2 Default Setting Click on the Restore button. Then, the system will restore factory default setting and automatically restart again. Changed network and SIP setting will be removed. 4.10 Reboot Press the reboot button. The system will restart automatically.
5. IVR / Keypad Management You can use the PSTN phone keypad to operate the IP Telephony gateway. Please follow the instruction to configure.
Use the * (star) key when entering a decimal point. Enter value-using numbers on the telephone keypad. Use the * (star) key when entering a decimal point. Enter IP address using numbers on the telephone keypad. Use the * (star) key when entering a decimal point. Enter IP address using numbers on the telephone keypad. Use the * (star) key when entering a decimal point. 1:G.711 u-Law, 2: G.711 a-Law, 3: G.723.1, 4: G.729a, 5: G.726 16K, 6: G.726 24K, 7: G.726 32K, 8: G.
6. Specification Model SP5001C SP5001D Standard Telephone Port SP5002A SP5012A IETF SIP (RFC3261) 1 FXS 1 FXS 2 FXS 1 FXS 1 PSTN (lifeline) Ethernet Port 2 10/100M ports for WAN and LAN connection Voice Codec: 1 FXO ● G.711: 64k bit/s (PCM) ● G.726: 16k / 24k / 32k / 40k bit/s (ADPCM) ● G.729A: 8k bit/s (CS-ACELP) ● G.729B: adds VAD & CNG to G.729 CNG, EC (G.
