PURE M3700, M3716, M3708, M3704, M3702 User Manual

...
M3700
Pure SIP Gateway
User Manual
Version: 11.0
Firmware: 1.10
DCC NO. 91710015011
M3700 user manual
Contents
1. Safety Instructions.................................................................................. 7
2. Preface..................................................................................................... 7
2.1 What is SIP..........................................................................................................7
2.1.1 SIP Clients..............................................................................................................7
2.1.2 SIP Servers ............................................................................................................ 8
3. Package Contents................................................................................... 9
4. Panel Descriptions ................................................................................. 9
4.1 Front Panel..........................................................................................................9
4.2 Rear Panel.........................................................................................................10
4.3 LED Indicators...................................................................................................11
4.4 Connectors ........................................................................................................12
4.4.1 Connect Console Port .......................................................................................... 12
4.5 IDC Connectors (Only for 3708/3716) ...............................................................13
5. Information required before Installation............................................. 14
5.1 IP Address .........................................................................................................14
5.2 SIP Information..................................................................................................15
5.3 Prepare a password for Web Management .......................................................15
6. Installation and Configuration............................................................. 15
6.1 Confirming the Region ID ..................................................................................16
6.1.1 Phone Setting....................................................................................................... 16
6.1.2 System console settings (Only 3704/3708/3716)................................................. 17
6.2 IP Address Settings ...........................................................................................17
6.2.1 Static IP Mode...................................................................................................... 18
6.2.2 DHCP Mode ......................................................................................................... 19
6.2.3 PPPoE Mode........................................................................................................ 19
7. SIP Configuration.................................................................................. 25
7.1 Channels and SIP entity....................................................................................26
7.2 SIP Proxy and Registrar Parameters.................................................................27
7.3 SIP Entity...........................................................................................................28
7.4 SIP Outbound Authentication.............................................................................28
7.5 Configure STUN for Client under NAT...............................................................29
7.6 Check SIP entity Status .....................................................................................31
7.7 Phone Book.......................................................................................................31
7.7.1 General Phone Book............................................................................................ 31
7.7.2 Hotline Function.................................................................................................... 32
7.8 Make SIP Calls..................................................................................................35
7.9 Make Inbound Transit Call.................................................................................36
7.10 Make SIP IP Call without SIP Proxy...................................................................37
8. Other Parameters.................................................................................. 38
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8.1 Dialing Plan........................................................................................................38
8.1.1 Dialing Method...................................................................................................... 38
8.1.2 Dial In Rewriting Rule........................................................................................... 41
8.2 Call Forward ......................................................................................................42
8.3 Inbound Authentication......................................................................................43
8.4 FAX.................................................................................................................... 43
8.5 Non-SIP Call port seizure preference................................................................44
8.6 Call Waiting........................................................................................................45
8.7 Target the Media (RTP)......................................................................................47
9. WEB MANAGEMENT INTERFACE....................................................... 48
9.1 BASIC / GENERAL............................................................................................49
9.2 IP SETTING.......................................................................................................51
9.3 ADVANCED / GENERAL...................................................................................53
9.4 SIP COMMON...................................................................................................55
9.5 SIP OUTBOUND AUTHENTICATION................................................................59
9.6 SIP INBOUND ANTHENTICATION....................................................................60
9.7 DIALING PLAN..................................................................................................62
9.8 FILE TEMPLATE................................................................................................64
9.8.1 Template of MEM file............................................................................................ 65
9.8.2 Related Configuration at Web Page..................................................................... 67
9.9 INBOUND TRANSIT..........................................................................................73
9.10 STUN.................................................................................................................75
9.11 CHANNEL .........................................................................................................77
9.12 PHONE BOOK...................................................................................................80
10. Use Private IP (Behind NAT) ................................................................ 81
11. File Management................................................................................... 82
11.1 File Types ..........................................................................................................82
11.2 Software Update................................................................................................82
11.2.1 Software update via FTP...................................................................................... 82
12. Appendix ............................................................................................... 85
12.1 Appendix A: Phone-Set Command....................................................................85
12.2 Appendix B: Console Command........................................................................87
12.3 Specifications.....................................................................................................88
12.4 Mapping table of characters used in PPPoE......................................................89
1 2.5 Region ID...........................................................................................................90
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Change History: Software Version 1.10
Incoming call to FXO can be forwarded
Incoming call can be forward to other SIP destination. Please refer to
9.11 CHANNEL
Add digit map function Digit map controls the call route from FXS port. Please refer to 8.1
Dialing Plan
RFC2833 is able to send Flash DTMF for G.723 Codec
This function is workable at this version
Support Min-Expire time from server for registration
If the default expire time is shorter than Min-Expire time of Proxy Server and server reply correct Min-Expire time message, M3700 adjust the Min-Expire time itself and register to SIP Proxy again.
SIP Entity registration control
Register or De-register all SIP entity. Please refer to 9.4 SIP COMMON
Register or De-register SIP Entity registration by phone-set
As left, please refer to
12.1 Appendix A: Phone-Set Command
Change History: Software Version 1.09
Voice Jitter is adjustable Jitter buffer is adjustable
Register expire time is adjustable
Configure Register expire time to insure registration status
Console command delete nvram can keep original setting
"delete nvram region_specific" keep all original setting except tone, ring and Time zone
Change History: Software Version 1.08
MEM file configuration Configuration can be backup and updated by MEM file
Support signal receiving Support signal receiving of 2833 DTMF for Codec-723
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of 2833 DTMF for Codec-723
Change default value of RFC 2833 DTMF
Default value of RFC 2833 DTMF is changed from Disable to Negotiate
Change History: Software Version 1.07
Hot Line support M3700 is connected to M4600B SIP Line or other SIP device
automatically when calling side picks up phone
Change History: Software Version 1.06
NAT signaling keep alive function
M3700 can send Dummy Packet to SIP Proxy to insure the connection is alive if the connection need to pass through NAT.
Contact information control for RTP packet to pass through NAT
M3700 can send RTP packet to contact information (IP, Port) according to Symmetric RTP or SDP in packet in order to improve the connection to pass through NAT.
Pass NAT environment without STUN server
System can specify NA T WAN IP when the connection to STUN server is disabled
Add new dialing plan control
Digits dialed from M3700 can be rewrite to different digits and sent to SIP Proxy.
Add Country Region ID Add Country Region ID for some country never listed in this manual
before
Correct error description in manual
Modify description of FAX by using SIP-based T.38 and G.711 codec. Modify description of Phone Book call with/without Proxy
Change History: Software Version 1.05
Default Public Address is generated automatically
Default Public Address is created by account username and Registrar
Default contact address is Contact address become read-only and it is the same as username
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generated automatically
Add SIP-based T.38 protocol
Standard SIP-based T.38 is applied. T.38 Proprietary is not retained
Add Route control Default route call to outbound proxy if outbound proxy is enabled and
Registrar is disabled.
Change History: Software Version 1.04
Creates default Realm for registration
System creates default Realm for authentication if users does not know the Realm
Add call waiting function Call waiting function for a FXS port to answer two calls
Change History: Software Version 1.03
Add new Time Configuration server
The gateway is able to receive time data from NTP Server as time synchronization source.
FAX function update T.38 is disabled if "0" is entered for Signal port number
Add incoming call screening function
System is able to block direct call to this gateway. Normal call is pass through the proxy server to this gateway.
Add CLIR function System is able to support Calling Line Identification Restriction
Offnet Call test is successful for Cisco model
Offnet Call to PSTN test is successful for Cisco AS 5350
Change History: Software Version 1.02
New function for Channels 1. Battery Reverse
2. Auto Answer
3. Directional control for FXO and FXS
4. Join SIP Entity and Connect Device change are available for both FXS and FXO
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New Auto Answer function for Inbound transit call to SIP
Disable Auto Answer, Auto Answer without PIN code, Auto Answer with PIN code for inbound transit call control.
Interpret T.38 and G.711 FAX operation
Explain the timing for T.38 and G.711 using in products
Interpret port seizure preference for VOICE/FAX on FXO and FXS port.
When make a non-SIP call, those preference will decide which port that the call will go.
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1. Safety Instructions
WARNING
1. Do not attempt to service the product yourself. Any servicing of this product should be referred to qualified service personnel.
2. To avoid electric shock, do not put your finger, pin, wire, or any other metal objects into vents and gaps.
3. To avoid accident al fire or electric shock, do not twist power cord or place it under heavy object s.
4. The product should be connected to a power supply of the type described in the operating instructions or as marked on the product.
5. To avoid hazard to children, dispose of the product’s plastic packaging carefully.
6. The phone line should always be connected to the LINE connector. It should not be connected to the PHONE connector as it may cause damage to the product.
7. Please read all the instructions before using this product.
Notice: The installation of M3700 is easy and quickly. Most of setting is pre-configured. Please read M3700 Quick Installation Guide for installation first. If you have further configuration, you can refer to this manual.
2. Preface
The M3700 unit is a personal SIP VoIP gateway developed using the latest in VoIP technology. It is also very simple to install and easy to operate.
2.1 What is SIP
2.1.1 SIP Clients
SIP clients include the following: (1) SIP Softphone: SIP client Software that runs at PC. It support SIP standard and can register to SIP Proxy for making calls. (2) SIP Gateway: SIP client Software that runs at a box. It support SIP standard and can register to SIP Proxy. General phone-set that connect to this box can make SIP IP call. (3) SIP IP Phone: SIP client Software that runs at a device that looks like general Phone-set. It support SIP standard and can register to SIP Proxy for making calls as using general phone-set. (3) SIP Wi-Fi Phone: SIP client Software that runs at portable phone with wireless LAN connection.
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It support SIP standard and can register to SIP Proxy. If wireless LAN connection keeps, the Phone can make calls in certain range without wiring.
2.1.2 SIP Servers
SIP servers include the following: (1) Proxy server—The proxy server is an intermediate device that receives SIP requests from a client and then forwards the requests on the client's behalf. Basically, proxy servers receive SIP messages and forward them to the next SIP server in the network. Proxy servers can provide functions such as authentication, authorization, network access control, routing, reliable request retransmission, and security. (2) Redirect server—Provides the client with information about the next hop or hops that a message should take, then the client contacts the next hop server or UAS directly. (3) Registrar server—Processes requests from UACs for registration of their current location. Registrar servers are often co-located with a redirect or proxy server. Hint: For most of ITSP (Internet Telephony Service Provider), the address (domain) of the servers above are consistent.
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3. Package Contents
The M3700 Gateway X 1 Power Core X 1 Accessories for fixing support X 1 (For 3708/3716) System CD-ROM X 1 5 IDC Connector X 4 (For 3708/3716) Rubber footer RJ-45 Ethernet Cable X 1 RJ-1 1 Telephone Cable X 1
4. Panel Descriptions
4.1 Front Panel
REGISTERED STUN
M3716 Front Panel
REGISTERED STUN
M3708 Front Panel
M3704 Front Panel
M3702 Front Panel
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4.2 Rear Panel
There is a button on the rear panel of gateway for special maintenance. Please don’t touch this button under normal operation.
M3716 Rear Panel
M3708 Rear Panel
M3704A Rear Panel
M3704B Rear Panel
M3704C Rear Panel
M3704D Rear Panel
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M3702A Rear Panel
M3702B Rear Panel
4.3 LED Indicators
LED Label Description
On Link up Off Link down
LNK/ACT
Flash Sending/Receiving data packets On (LNK is on) 100Mbps
10/100 Ethernet
100Mbps
Off (LNK is on) 10Mbps On Off hook Off On hook
FXS
Flash Ringing out On Line is active Off Line is inactive
LOOP/RING
FXO
Flash Ringing in
Alarm The red light “On” indicates that system has some
problem; please contact your vender.
Power “On” indicates that the power supply is working
normally. CPU/ACT “On” indicates that the CPU is working normally. Registered “On” indicates that all SIP entities are registered
successful.
“Off” indicates that all SIP entities are registered fail.
“Flash” indicates that at least one of these SIP entities is
registered fail.
Device
STUN “On” indicates communicate with STUN Server once.
“Off” indicates never communicate with STUN Server.
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4.4 Connectors
Ports Label Description
FXS Connects to a telephone set or fax
machine
Voice Ports
FXO Connects to the phone line LAN/Internet RJ-45 connector
MDI-X connects to a Modem
Ethernet Ports
PC RJ-45 connector
MDI connects to a PC Console Port (Only 3704/3708/3716)
Console RJ-45 connector/RS-232 Interface
4.4.1 Connect Console Port
To connect port, connect the PC with this machine via RS-232 Console cable, power on the PC and configure the PC parameters as following:
 Speed: 9600  Data Bits: 8  Parity Check: None  Stop Bit: 1  Flow Control: None
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If Console cable is not available, run Telnet in PC for connection is OK.
4.5 IDC Connectors (Only for 3708/3716)
IDC connector is used for the voice interface (FXS and FXO) on the frame model. IDC connector can easily connect PBX line and telephone wire together to the gateway. No special tools are required; please follow the instruction to install: (Remarks: For IDC connector, it’s better to use No. 24 wire, e.g. CAT 5)
Get the material ready
Insert the insulated wires directly into the
block for wire insertion
Push the block down until it is locked to
flush the conductor with the probe
Cut off the conductor outside the edge to avoid from causing the circuit shortage
Push from here
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5. Information required before Installation
You need to prepare the following information before installing the gateway.
5.1 IP Address
The gateway requires an IP address for operation. Before installation you need to know how to obtain an IP address from your local ISP. Static IP, DHCP or PPPoE can be used. The following table helps you to decide what information you need. If your ISP offers static IP, you may need to obtain an IP from MIS personnel in order to prevent an IP conflict. Otherwise DHCP (most cable broadband providers offer this) and PPPoE (most ADSL broadband providers offer this) will work fine.
IP Environment Requiring information
Public IP Address
IP Address Subnet Mask
Default Gateway
It is strongly suggested that you obtain an IP address from MIS personnel in order to prevent an IP conflict.
Static IP
Private IP Address
IP Address Subnet Mask Default Gateway It is strongly suggested that you obtain an IP address from MIS personnel in order to prevent IP conflicts. Your private IP requires an IP Sharing device and you must configure the IP Sharing device to treat the unit and the IP
that it is using as a virtual server. Dynamic IP address (DHCP) DHCP mode PPPoE Account Number
Password
Your ISP normally provides this information.
If you don’t have this information please
contact your ISP.
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5.2 SIP Information
Before configuring SIP, the M3700 requires SIP information for operation. The following table helps you to decide what information you need.
Items Description
1. SIP Proxy If you want to make SIP calls through SIP proxy
server, you will need to know the IP address or domain name of SIP proxy server. The proxy server is an intermediate device that receives SIP requests from a client and then forwards the requests on the client's behalf. If you don’t know which SIP proxy for setting, contact your SIP service provider.
2. Public Address (SIP Account) Example: sip@abc.com
The public address is like phone number, you can get the account from your SIP service provider.
3. Outbound Authentication You will need the information when the SIP
proxy server requires authentication. You can get this authentication information from SIP service provider when you apply for the service.
5.3 Prepare a password for Web Management
You will need to prepare a password for Web based Management. It can be a digit and/or letter combination ranging from 1 to 6 digits (E.g. 123). For security reason, password must be set to enter the Web Management page.
6. Installation and Configuration
After preparing the information you need as specified in section 5, follow the following steps to do the basic configuration. You can use either a telephone or a system console to perform basic configurations. It is simple to connect a telephone set to FXS port and configures the system. If you want to use system console to configure the system (Only 3704/3708/3716 support), you have to configure your VT100 terminal to match the settings of the unit’s console port. The console port’s terminal connection is set to 9600 baud, 8 data bits, 1 stop bit and no parity. Turn on the unit’s power and wait for the terminal to display “Press Enter…” follow the directions to begin.
Here are several procedures to do:
1. Confirming the Region ID.
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2. Configure IP address of gateway.
3. Enter into the WEB page.
4. Plan and configure the channels into SIP entity. (if needs. By default, all channels joins Entity 1)
5. Configure SIP proxy and registrar information.
6. Configure SIP entity information.
7. Configure Outbound Authentication (If needs).
8. Configure STUN (If your gateway is behind NAT).
9. Check the SIP entity if is registered successful.
10. Configure Phone book (If needs)
11. Make a SIP call.
6.1 Confirming the Region ID
Skip this step if you are installing your unit in the default region. The default Region ID is printed on the label located outside the box. If you are installing your unit at any region other then the reg ion ID specified on the label, you will then need to configure the unit to the correct Region ID. About the Region ID, please refer to Section
12.5 Region ID.
Region ID Label
M3702
6.1.1 Phone Setting
1. Connect the power.
2. Connect the phone cable to the “Phone” socket on the rear panel as pictured above.
3. When the CPU/ACT LED is on, pick up the handset and listen for the dialing tone.
4. Dial “##0000” and listen for 3 short beep.
5. Dial “9507
#”Assuming you are modifying for China (The last 2 digits are the regional ID)
6. Dial “971
#” ;Sets the new regional ID.
7. Hang up the phone. The device will be updated with the new region setting after it restarts (restart time is about 10 seconds)
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6.1.2 System console settings (Only 3704/3708/3716)
The Instructions below is to set Region ID from Console terminal or Telnet (For Telnet, From Windows, Start Æ Run Æ Telnet 192.168.0.2 (192.168.0.2 is the default IP of this
machine) (In the example, the Region ID is changed to 07 for China)
SIP-RG>enable SIP-RG #configure Enter configuration commands, one per line. End with CNTL/Z SIP-RG(config)#regional_id 07 SIP-RG(config)#exit SIP-RG#delete nvram region_specific This command resets the system with factory defaults. All system parameters will revert to their default factory settings. All static and dynamic addresses will be removed. Reset system with factory defaults, [Y]es or [N]o? Yes
Note: Here is some related command for your reference The following instruction reset all setting: “delete nvram” The following instruction reset all setting except IP address: “delete nvram keep_ip”
6.2 IP Address Settings
We recommend using a traditional phone to configure the unit’s parameters, as this is the easiest way. The following two sections contain the procedures used to configure the unit according to how you obtain your IP address (Static IP; DHCP or PPPoE).
Every time you set a parameter item and press the “#” key to complete it, a successful setting will be confirmed by three equal tones in succession. If your setting is unsuccessful you will be prompted with one long tone.
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6.2.1 Static IP Mode
The following table shows an example. IP Address 210.67.96.121 Subnet Mask 255.255.255.248 Default Gateway 210.67.96.120 Web Management Password
123
Using the information contained in the example above. The procedure is as follows:
1. Connect the unit to a suitable Power source.
2. Connect a traditional phone set to the “FXS” connector located on the rear panel.
3. When the CPU/ACT light is on, pick up the phone to hear the dialing tone.
4. ##0000 ; you should hear three short tones.
5. 010
# ; the digit “0” is used to enable “manual” IP mode.
6. 02210*67*96*121
# ; IP address.
7. 03255*255*255*248
# ; Subnet Mask.
8. 04210*67*96*120
# ; Default Gateway.
9. 15123
# ; “123” is the web management password.
10. 981
# ; Warm-restarts.
11. Hang up the phone. The system should now restart.
You can also use console to configure IP address. (For Telnet, From Windows, Start Æ Run Æ Telnet 192.168.0.2 (192.168.0.2 is the default IP of this
machine.)
SIP-RG>enable SIP-RG#configure Enter configuration commands, one per line. End with CNTL/Z SIP-RG(config)#ip state user SIP-RG(config)#ip address 210.67.96.121 255.255.255.248 System need to restart SIP-RG(config)#ip default-gateway 210.67.96.120 SIP-RG(config)#exit SIP-RG#restart This command resets the system. System will restart operation code agent. Reset system, [Y]es or [N]o? Yes
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6.2.2 DHCP Mode
1. Connect the unit to a suitable Power source.
2. Connect a traditional phone set to the “FXS” connector located on the rear panel.
3. When the CPU/ACT light is on, pick up the phone to hear the dialing tone.
4. ##0000 ; you should hear three short tones.
5. 011
# ; the digit “0” is used to enable “DHCP” IP mode.
6. 15123
# ; “123” is the web management password.
7. 981
# ; Warm-restarts.
8. Hang up the phone. The system should now restart.
You can also use console to configure IP address. (For Telnet, From Windows, Start Æ Run Æ Telnet 192.168.0.2 (192.168.0.2 is the default IP of this
machine, change it if the IP is changed.)
SIP-RG>enable SIP-RG#configure Enter configuration commands, one per line. End with CNTL/Z SIP-RG(config)#ip state dhcp SIP-RG(config)#exit SIP-RG#restart This command resets the system. System will restart operation code agent. Reset system, [Y]es or [N]o? Yes
6.2.3 PPPoE Mode
If your network environment is using PPPoE, you need to prepare the information as specified in section
5. Information required before Installation. The following table shows an example. PPPoE Account 83721@hinet.net PPPoE Password 123ab Web management password 123
There are three ways to configure user name and password of PPPoE
6.2.3.1 1. Use phone set to configure:
You can configure the user name and password by using phone set. The command ‘09’ is used for username and ‘10’ is for password of PPPoE. Since the user name and password use characters and digits are accepted by phoneset only, you need a mapping between characters and digits. You
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can find them at section 12.4 Mapping table of characters used in PPPoE. Example user name:83721@hinet.net,Password:123ab,The procedure is below
1. Connect the phone to the unit
2. When CPU/ACT is light, pick up the phone and press
3. ##0000
You will hear 3 short tones.
4. 0938333732314068696*465742*46*46574#
Set user name83721@hinet.net
5. 103132336162#
Set password is 123ab
6. 981#
Save and restart.
6.2.3.2 2. Use Console to configure (Only 3704/3708/3716)
You can also use console to configure IP address. (For Telnet, From Windows, Start Æ Run Æ Telnet 192.168.0.2 (192.168.0.2 is the default IP of this
machine, change it if the IP is changed.)
SIP-RG>enable SIP-RG#configure Enter configuration commands, one per line. End with CNTL/Z SIP-RG(config)#pppoe username 83721@hinet.net SIP-RG(config)#pppoe password 123ab SIP-RG(config)#exit SIP-RG#restart This command resets the system. System will restart operation code agent. Reset system, [Y]es or [N]o? Yes
6.2.3.3 3. Use WEB Interface to configure:
You can configure the user name and password by using WEB interface. Follow the steps to finish configuration.
Step 1: Using a traditional phone set to configure the web management password and phone number You will need to use a web browser to perform the PPPoE settings through the unit’s web based management interface. To enter the web based management interface you must have a previously configured password. Follow the next procedure to setup your password and phone number.
1. Connect the unit to a suitable Power source.
2. Connect a traditional phone set to the “Phone” connector located on the rear panel.
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3. When the CPU/ACT light is on, pick up the phone. You should hear the dialing tone.
4. ##0000 ; you should hear three short tones.
5. 15123
; “123” is the web management password.
6. 010# ; “0” is to enable “manual” IP mode.
7. 02192*168*0*2# ; IP address.
8. 03255*255*255*0# ; Subnet Mask .
9. 981
# ; Used to restart the unit.
10. Hang up the phone to complete the configuration.
Step 2Configure IP address of PC
Use the provided Ethernet cable to connect your PC to the port labeled “PC”, located on the rear panel of the unit. For M3704, 3708, and 3716, it is located on the front panel.
Because the unit’s default IP setting is 192.168.0. 2, you must configure your PC to the same subne t. “192.168.0.x” for example. The following example uses 192.168.0.5 for the IP address and
255.255.255.0 for the subnet mask.
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After you have completed the PC’s IP address setting, you will be required to rest art the PC in order for the new settings to take effect.
Step 3: Using the browser to configure the PPPoE Parameters of the gateway. On the PC that is connected to the unit, enter the unit’s IP address (Default 192.168.0.2) and press enter. The unit will then prompt you with a dialogue box requesting that you enter a password. Use “WEB” (all capitals), for the User field and “123” for the password field that you have previously configured (Please keeps password blank if you don't configure password "123" before). Click the OK button; you should now have access to the unit’s web based management interface page.
The unit’s IP address
“WEB” should be all Capitals
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6.2.3.4 Upon entering the web based configuration interface.
Click on “IP SETTING” at the top of the page and you will see the page as shown in the following image. Select PPPoE from the “IP State” pull down menu. Fill in the “Account”, “Password”, and “Confirm Password” under the PPPoE Settings. You can obtain this information from your ISP. Click on the Apply button. Click the “BASIC” button at the top to go to the BASIC page and select “Warm Start” to restart the gateway. You can also perform a warm start using the phone by picking up the handset and dialing “##0000” then “981#”. After restarting, the gateway will use PPPoE to obtain it’s IP address. Web Folder: IP SETTING\
1
Click “IP setting” to open this display
2
Click the “Apply” button to apply any changes.
3
4
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Web Folder: BASIC\
5
6
Click the “Apply” button to apply any changes.
At this stage, your unit should be able to use PPPoE to access the Internet. However, if you configured a wrong account number or password, your unit cannot access the Internet. You are not able to use PC to access unit by using the IP address of 192.168.0.2 because unit has been set in PPPoE mode. You have to use phone set to configure unit back to fix IP mode (##0000 010#) and use PC browser to configure correct parameters.
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7. SIP Configuration
M3700 not only can make regular PSTN calls, it also can communicate with IP Phones or Soft-Phones by using SIP protocol. This section shows you what parameters you need to configure for SIP calls and how to make the SIP calls.
Notice: These configurations on WEB page, after select or input value in the field, please press “Apply” button to save and confirm the setting. Some parameters need “Warm-restart”, please process the restart action, thanks.
M3702/3704 (SIP)
SoftPhone (Notebook
M3704
IP
Cisco IP Phone
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7.1 Channels and SIP entity
Many Channels can be assigned as one SIP Entity. Single Channel also can be assign as one SIP Entity.
Application example: As the figure below, Channel 1-3 belongs to SIP Entity 1:
001@abc.com. Channel 4 and Channel 5
belongs to SIP Entity 2:
002@abc.com. and Channel 6-8 belongs to SIP Entity 3: 003@abc.com.
When other device under SIP network dial into
001@abc.com, the phone connect to Channel 1 is
ringing. If Channel 1 is under conversation (busy), the line will be switched to Channel 2, and so on. So Channel 1~3 become a simple Hunting Group. (This feature needs the support of SIP Proxy Server).
Figure:
M3708
FXS
Internet
Busy
SIP IP Phone
Ring
Configuration: WEB page: CHANNEL\
26
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