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M3700

Pure SIP Gateway

User Manual

Version: 11.0

Firmware: 1.10

Update: 2006/05/12

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

M3700 user manual

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

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

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

M3700 user manual

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.

M3700 user manual

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.

M3700 user manual

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

M3700 user manual

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

M3700 user manual

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.

M3700 user manual

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)

M3700 user manual

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

M3700 user manual

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 name：83721@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.

M3700 user manual

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 2：Configure 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

M3700 user manual

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\

Click “IP setting” to open this display

Click the “Apply” button to apply any changes.

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Web Folder: BASIC\

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.

M3700 user manual

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

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\

M3700 user manual

Notice: Each channel must belong to a SIP entity.

7.2 SIP Proxy and Registrar Parameters

You need to configure IP address or Domain name of Registrar and Outbound Proxy server (optional), please check the information is right. SIP service provider will give you an IP address or Domain name of Registrar and Outbound proxy when you apply for the service.

Configuration WEB Page: ADVANCED\SIP COMMOM

Notice: Generally speaking, Registrar and Outbound Proxy server is same. Fill in the domain name according to your SIP Proxy Provider.

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7.3 SIP Entity

SIP service provider will assign one or more SIP accounts for you when you apply for the service. In standard, the SIP account is called ‘Public Address’, so you need to configure the account information in ‘Public Address’ item. The format is like an E-mail address such as

mary@abc.com.

The Public Address will be generated automatically with the format below if user keeps the Public Address empty. "Default account's username" @ "Registrar" if you had enter the information below

1. Registrar Setting. For example: fwd.pulver.com, which configured at

7.2 SIP Proxy and

Registrar Parameters

2. Username of Default Account. For example: 413189, which is configured at below graph For example: If the two data above is created, then the Public Address will be 413189@ fwd.pulver.com

Input Username and Password here if SIP Proxy needs it for authentication. This account information also helps you to create Realm for SIP Outbound Authentication and Public Address.

Configuration

WEB Page: ADVANCED \ SIP COMMON

You can control the SIP entity on WEB page, just select ‘Enable’ or ‘Disable’.

7.4 SIP Outbound Authentication

You need to configure outbound authentication for each SIP entity if SIP proxy server or other SIP

phone request for authentication. Please check with SIP service provider if you need the setting.

Please select the entity then input information includes realm, username, and password.

M3700 user manual

"Realm" is a kind of verification for SIP Outbound Authentication. If SIP service provider does not provides this information. The gateway will create a default Realm (by string USER-UNSPECIFIED-REALM) automatically with your Username and Password mentioned on last section for SIP Outbound Authentication. If there are more than one SIP entity is registered on this gateway. The gateway creates Realm for each entity. The default Realm helps you to register the SIP server successfully.

Configuration WEB Page: ADVANCED \ SIP OUTBOUND AUTHENTICATION

7.5 Configure STUN for Client under NAT

STUN is an application-layer protocol that can determine the public IP Ad dress of a NAT device that sits between the STUN client (M3700) and STUN server.

Notice:

1. If your gateway is behind NAT (Use Private IP), please consult the SIP service provider to

provide information of STUN server and also configure the parameter here, otherwise you need to input NAT WAN IP to penetrate NAT device Please refer to section

9.10 STUN after

configuring the parameters of STUN, please act Warm-Restart.

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2. If no useable free STUN Server available, for most of ITSP (Internet Telephony Service

Provider), their Outbound Proxy Server supports Media Relay, device under NAT can penetrate NAT without configuration.

3. STUN does not support Symmetric NAT.

STUN Server

Internet

M3700

NAT Device (Router, IP sharing device)

NAT WAN IP

Configuration

WEB Page: ADVANCED\STUN

You can enable and disable the service on WEB page.

M3700 user manual

7.6 Check SIP entity Status

You can use the WEB page to check the SIP entity is registered successful or unsuccessful.

WEB Page: ADVANCED\SIP COMMOM

If the status shows “REGISTERED” means successful, otherwise means fail; please notice that.

When you find the registration is fail, first check the “Registrar Setting” configuration is normal, or not “Enable”. Then check the “Public Address” and “Outbound Authentication” configuration is in normal status. If the configurations are all right, please check the situation with your SIP service provider.

7.7 Phone Book

7.7.1 General Phone Book

Since the SIP phone number is not easy for regular phone to dial, M3700 provide a SIP phone b ook to let standard phone to make a SIP call easier. The phone book uses index number to map SIP account. User also can configure this index number to build the route by SIP Proxy or build the route without Proxy if destination gateway use fixed IP (Public IP or private IP in VPN) For instance if the phone book is configure as below:

Index Public Address Port Via Proxy

100 01@61.220.145.70 5060 No <-- GW1 200 73797@fwd.pulver.com 5060 Yes <-- GW2 201 73797@61.222.217.5 5060 No <-- GW2

Notice: If your SIP account is digit type like

234@SIP.abc.com or 456@SIP.abc.com, and this

M3700 is register to SIP proxy: SIP.abc.com, you don’t need to configure the items.

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Configuration WEB page: PHONEBOOK \

7.7.2 Hotline Function

A new Hotline function is added for M3700 Firmware Version 1.07 or above When hotline function is enabled, the FXS channel is connected to specified SIP device or M4600B SIP Line (if the M3700 is configured and register to M4600B SIP Line as a client) automatically when user of M3700 FXS channel picks up hand-set. ♦ If the FXS channel is Hotlined to other SIP device (SIP Phone, Softphone), other SIP device

rings immediately when FXS channel user of M3700 picks up hand-set.

♦ If the FXS channel is Hotlined to M4600 SIP Line, (skip this section if the M3700 don't register to

M4600 SIP Line) FXS channel user of M3700 hear dialing tone from M4600B SIP line when pick up hand-set, and then he/she can dial extension number to other SIP device or Outbound Call to PSTN via IP-PBX environment.

Configuration of Hotline

♦ Enable Hotline function WEB page: PHONEBOOK \

♦ Setup index number WEB page: PHONEBOOK \

When Hotline function is enabled, user also needs to specify which channels (FXS only) should join Hotline function and which SIP number (Public Address) the channel is hotlined to.

M3700 user manual

Hotline mapping table

Channel (FXS) only Index Number Description

1st FXS channel 1 Index number “1” maps the 1st FXS channel 2nd FXS channel 2 Index number “2” maps the 2nd FXS channel …. …. … 16th FXS channel 16 Index number “16” maps the 16th FXS channel

Available Hotline index number

Model Available Hotline Index Number Note

M3702A 1 M3702B 1, 2 M3704A 1, 2 M3704B 1, 2, 3, 4 M3704C None No FXS channel is available M3704D 1, 2, 3 M3708 Depends on module used. Please refer to

table below.

Only FXS channel can be counted as index number

M3716 Depends on module used. Please refer to

table below.

Only FXS channel can be counted as index number

M3708/M3716 channel mapping number

Model

Group Location Channel Number (Please

select FXS port only)

Group 1

Lower module(S1), 4 ports of left side

1 2 3 4

Group 2

Lower module(S1), 4 ports of right side

5 6 7 8

Group 3

Upper module(S2), 4 ports of left side

9 10 11 12

3716

Group 4

Upper module(S2), 4 ports of right side

13 14 15 16

Group 1 4 ports from left 1 2 3 4

3708

Group 2 4 ports from right 5 6 7 8

Any index number that is not listed in Available Hotline Index Number is recognized as normal index number and they are not used as hotline function and not all of the channels have to join hotline function. Please see the example below Example Model: M3704B

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Index Public Address Port Via Proxy Description

1 01@61.220.145.70 5060 No

Channel 1 Hotline to

01@61.220.145.70 without

proxy

2 73797@fwd.pulver.com 5060 Yes

Channel 2 Hotline to 73797@fwd.pulver.com by proxy,

100 jack@fwd.pulver.com 5060 Yes 200 mike@fwd.pulver.com 5060 Yes 300 Jason@fwd.pulver.com 5060 Yes

No hotline function for channel 3, 4 to dial

User of 1st FXS channel picks up hand set, and then 01@61.220.145.70 rings immediately User of 2

FXS channel picks up hand set, and then 73797@fwd.pulver.com rings immediately

Hotline to M4600B SIP Line (skip this section if you don't register to M4600 SIP Line)

Assume the Public Address of M4600B SIP Line is

1234567@61.220.145.70 and it has extension

number 1001 to 1002.

So we configure the Phone Book as below

Index Public Address Port Via Proxy Description

1 1234567@61.220.145.70 5060 Yes

Channel Hotline to 1234567@61.220.145.70 SIP Line directly

2 1234567@61.220.145.70 5060 Yes

Channel Hotline to 1234567@61.220.145.70 SIP Line directly

User hears dial tone from M4600B SIP Line when pick up hand set and then dial extension no. for example 1002, to other SIP device

the unit SIP Line Entity: 1234567@61.22

0.145.70

M4600 M3700

SIP

SIP Phone (Notebook)

Hotline to M4600B SIP Line

1001

1002

M3700 user manual

7.8 Make SIP Calls

After you have configured the SIP phone on the SIP phone book, you can easily make SIP calls.

You can select one way to make SIP call following these ways:

Standard Call: Dial <numbers>+<#>.

1. Compare dialing plan, check the number if it is in setting. Example 050.

2. If the number is in setting, send the call to proxy. If the calls does not match dialing plan or the registration to the proxy is fail, then the call will be sent to PSTN.

3. If the number is not in dialing plan, the call will be sent to PSTN.

Phone Book Call: Dial <#>+ <index>+<#>.

1. Compare SIP Phone books; check the number if it is in phone book.

2. If the number is configured in Phone Book and Proxy selection is set to "No", you will hear a busy tone. If Proxy selection is set to "Yes", then send the call to proxy.

3. If the index number you had configured to use Via Proxy but it communicates with proxy failed, you will hear busy tone.

4. If the number is not in phone book, you will hear busy tone.

Force PSTN Call: Dial <*>+<numbers>. Always go through PSTN

Hotline Call: If the channel is configured to use Hotline function, any dialing above is disabled. If the channel is hotlined to other SIP device, no dialing is needs after user picks up handset. Other SIP device rings immediately.

Hotline Call to M4600B SIP Line:

Dial <SIP extension number> or

1. If you dial SIP extension number, other SIP device that register to M4600 SIP Line with that

SIP extension number will ring.

2. If you dial Prefix number, the call is relay to the IP-PBX network according to the Prefix Map

specified in M4600 SIP Line.

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7.9 Make Inbound Transit Call

To make an inbound transit call from PSTN to SIP, you have to enable Auto Answer function of this gateway

Please enable Auto Answer configuration at

WEB Page: CHANNEL

WEB Page: ADVANCED\GENERAL

Notice: If you do not want to dial “#” after numbers, please configure the ‘Dial Ending Time’ item. After the seconds configured here, the call will be sent automatically.

If you don't enable the Auto Answer configuration, the inbound call from PSTN will be assigned to a free FXS port of this gateway directly. It makes Inbound Transit Call impossible.

When Auto Answer function is enabled, the gateway will answer the call and calling side will hear the second dial tone. For the Auto Answer function, it is also divided into Enable and Enable w/

Pincode options. The configuration page is the same as above.

Dial Inbound Transit Call when Auto Answer is configured as Enable

Please dial the number below after the second dial tone:

1. SIP Number + ‘#’, Example: 73797# or

2. ‘#’ + Index Number + ‘#’, Example: #123#

If you still need to make a call to the FXS port of this gateway, please press "*" to seize a free FXS port.

M3700 user manual

Dial Inbound Transit Call when Auto Answer is configured as Enable w/ PIN code

This Auto Answer mode provides security control for the Inbound Transit call

Please dial the number below after the second dial tone:

1. PIN code + ‘#’+ SIP Number + ‘#’, Example: 7742#73797# or

2. PIN code + ‘#’+ ‘#’ + Index Number + ‘#’, Example: 7742##123#

If you still need to make a call to the FXS port of this gateway, please press "*" to seize a free FXS port.

Notice for the Inbound Transit Call

1. If the SIP number that user dial does not match any prefix code configured in Dialing Plan page, the call is disconnected.

2. If the PIN Code does not match any passwords configured in Password For Inbound Transit page, the call is terminated.

3. If the Index Number does not match any pre-configured Phonebook Index in Phone Book page, the Index Number will be regarded as SIP number and create a IP call without applying any match rule configured in Dialing Plan.

For which free FXS port that this gateway will seize, please refer to

8.5 Non-SIP Call port seizure preference The PIN code (Password for Inbound Transit) is configured at chapter 9.9 INBOUND TRANSIT The Dialing Plan is configured at chapter

8.1.1.1 Dialing Plan

The Index Number is configured at chapter 9.12 PHONE BOOK

7.10 Make SIP IP Call without SIP Proxy

The main purpose of Contact Address is making SIP calls without proxy. The Contact Address is the same as the "Username" of Public Address if that field is configured. For S/W version above 1.05, the value is read only. Generally speaking, "Username" of Default Account are digits and it is regarded as SIP number. WEB Page: ADVANCED\SIP COMMOM

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Making SIP calls without proxy server: The SIP protocol allows you to make SIP calls directly to the destination number without through the proxy server. You can simply dial the SIP number to connect other SIP gateway. The typical example is:

413189@61.220.145.70. mailto: Other SIP gateway that had already configured 413189@61.220.145.70 in Phone Book can connect this gateway by number 413189 without

routing through SIP Proxy.

Notice: For this type of SIP calls, the destination device’s IP address is already known and fixed.

8. Other Parameters

8.1 Dialing Plan

Dialing Plan controls the dialing behavior of users

8.1.1 Dialing Method

According to different configuration, user needs to select different dialing method.

(1) Dialing Plan: When the first (few) digits that user dials matches Dialing Plan (described in

next section

8.1.1.1 Dialing Plan), number is send to SIP Proxy and build call route to SIP

device, otherwise, make call route via local FXO port.

(2) Transparent: All numbers user dials are sent to SIP proxy server and all number that

controls M3700 is disabled, including the end code # of each dialing.

(3) Transparent with digitmap: All numbers user dials are sent to SIP proxy server and if any

numbers match digitmap, number is send to SIP Proxy immediately without waiting dial end time. Please refer to

8.1.1.2 Digit Map

Web Folder: ADVANCED\DIALING PLAN

M3700 user manual

8.1.1.1 Dialing Plan

Entry "X" means all calls will be sent to SIP proxy, if the SIP call is fail, it is disconnected. Only if the registration to SIP Proxy is failed, then the gateway will try to connect the number by PSTN. Registraton of SIP Proxy can be configured on Web Page: SIP Common. Please refer to

9.4 SIP COMMON If the configuration is only ‘050’ means the numbers like 050xxxxx will send to SIP proxy, if you dial any other numbers like 100, the number will send to PSTN immediately.

M3704A

FXO

FXS

Dialing Plan: 050 and 070

Dial 82261234 The call is sent to PSTN immediately

Dial

050123456 070345678 or

The call will be defined as SIP call and sent to SIP Proxy. If the SIP call is fail, then it is disconnected.

Configuration WEB Page: ADVANCED\Dialing Plan

8.1.1.2 Digit Map

Advantage

1. Able to create usable rule in digit map that is convenience for user to make calls. By this digit map, M3700 can simulate the FXS port of M4600 Plus. It also makes dialing behavior more easily.

2. When digit map is enabled, the outgoing call that fits the rule goes immediately and wait dial ending time is not required.

Dialing Method

Before you start to use digit map, change dialing method is required. Change Dialing Method to

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Transparent (with Digitmap) Web Folder: ADVANCED\DIALING PLAN

Then you are able to use the transparent function with digitmap. Transparent means all numbers user dials are sent to SIP proxy server and all number that controls M3700 is disabled, including the end code # of each dialing. By this way , all numbers user dials compare with the digitmap. When the number dialed matches the digit map, the number send to Proxy Server immediately without waiting dial ending time.

By the example figure above, we configure some example of digitmap Here is the explanation of rule

(1) X means any digits (2) [ ] means the digits in the [ ] are all acceptable, such as [479] (3) [ ~ ] means the range between ~ are all accept able. For example, [2~4] means the number 2, 3,

4 are all acceptable

(4) "." means the previous digit can appear again. For example, "X." means 22, 33, 44... are all

acceptable.

Example Description *[389*]X.[#8] Number that match *+ 3 or 8 or 9 or * + any digit + repeat

M3700 user manual

previous digit + # or 8 are send to proxy server without waiting dial ending time

*2XX Number that match * +2 + any digit + any digit are send

to proxy server without waiting dial ending time

8.1.2 Dial In Rewriting Rule

Number dialed from M3700 can be converted to different number and sent to SIP Proxy. User can pre-define maximum 10 sets of prefix rewriting rule to convert the number that user dials before build the connection to SIP Proxy. It is useful to create a user-friendly dialing behavior and also can limit user to dial certain number. The rules below explain the judgment.

1. System will check the dialing plan on last page in advance to decide whether it is PSTN call or SIP call.

2. If the call will be send to SIP Proxy, then system will exams the number to see if it meets Rewriting Rule.

3. If the SIP call does not meets any Rewriting Rule, system will build the SIP call with the number that user dials.

4. If the numbers of the SIP call meets any Rewriting Rule, then the numbers is converted (or limited if it meets barring rule) and system build the SIP call by converted number.

Here is the example Web Folder: ADVANCED \ DIALING PLAN

Pattern: Add the pattern that user may dial Rewrite: Add the converted number if user dials the same digits in pattern column. Fill in digits and click the AddDialin button

By the operation above, we create a Rewriting Rule table below and it controls all SIP call. The example table below illustrate that all call are converted to the phone number that includes

DCC NO. 91710015011

Country Code + Area Code + Phone No., and then sent to proxy, and prefix phone number 0204 is forbidden.

Pattern Rewrite X means any digits. ! means the call is terminated.

00x

If the prefix number dials from user are 001~009, then the 3 digits are removed. For example, if user dials 0028621123456, then the system dials 86211123456 to build SIP call.

0 886

If the prefix number dials from user are 0, then the digit is replaced with 886. For example, if user dials 0921123456, then the system dials 886921123456 to build SIP call.

x 8862x

If the prefix number dials from user are 1~9, then add 8862 in front of the original number. For example, if user dials 82263368, then the system dials 886282263368 to built SIP call.

0204 !

If the prefix number dials from user are 0204, then the call is terminated.

Matching Rule

1. Best Match rule, the longest digits match first.

2. Wildcard ( x digits) match last

8.2 Call Forward

There are three forward types:

1. All: All incoming VoIP call to the SIP entity will be forward.

2. Busy: When the SIP entity is busy, the incoming VoIP call will be forward.

3. No Answer: When the SIP entity is no answer and after 30 seconds, the incoming VoIP call will be forward.

Notice:

 In order to let the caller identify the port has been configured ”forward”; the caller will hear

second dial tone, rather than normal dial tone.

 If Auto Answer function is disabled, incoming call from PSTN seizes a free FXS port. The call

is not forwarded even the seized FXS port is part of Call Forward SIP Entity.

M3700 user manual

 If Auto Answer function is enabled, Incoming PSTN call dials "*" to seize a free FXS port after

second dial tone. The call is not forwarded even the seized FXS port is part of Call Forward SIP Entity.

 If Auto Answer function is set to Forward to SIP, Incoming PSTN call is forward to new

destination configured in the entity that this channel belongs to.

Configuration WEB page: ADVANCED\SIP COMMOM

Phone Set: Please refer to section Appendix A: Phone-Set Command.

8.3 Inbound Authentication

You need to configure inbound authentication if you request authentication for other SIP phone to call you. Configuration WEB Page: ADVANCED \ SIP INBOUND AUTHENTICATION

8.4 FAX

For M3700 software version 1.05 or above, SIP-based T.38 Fax protocol is applied. Any brand SIP gateway with SIP-based T.38 Fax protocol may transmit FAX with each other. T.38 is FAX protocol and it has better performance and better successful transmission rate. However, SIP device that does not support SIP-based T.38 still can transmit and receive FAX with M3700 by G.711 codec. G.711 codec uses more bandwidth, so it may not as good as SIP-based T.38 protocol if bandwidth control is the key factor of the network.

Setup method is listed below:

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1. Web folder: “Channel” Enable T.38 Fax Relay support. Configure it to Yes

2. Warm-Restart the system

Note: For FAX transmission, two gateways will change to SIP-Based T.38 Protocol automatically if both sides support SIP-based T.38.

Note:

If M3700 connects different SIP devices, some have T.38, but some use G.711 codec only, then user should enable G.711 codec support for FAX. Setup method is listed below:

1. The same step as above set Connect Device to Fax

2. Setup “Codecs Type“, Web Folder: ADVANCED\SIP COMMON

Select and mark “PCMU” and “PCMA” Codecs (G.711 Standard), than click “Apply” button

3. Warm-Restart the system

8.5 Non-SIP Call port seizure preference

For non-SIP Calls, the port seizure preference is listed below

1. Inbound from PSTN

If the inbound FXO port was configured as "Fax" device, it will also seize only FXS ports that "Connect Device" is configured as Fax. The Voice devices behave the similar way.

From FXO port to FXS port Note

Connect Device at FXO port Connect Device at FXS port

M3700 user manual

VOICE port Select VOICE port only From the lowest port number

upward

FAX port Select FAX port only From the lowest port number

upward

2. Outbound to PSTN

For the calls from FXS to FXO, the ports of the same "Connect Device" type will be the prior selection for the calls.

If there is no correct configured port is available, it will ignore the "Connect Device" setting and create a call as the rule below.

From FXS port to FXO port Note

Connect Device at FXS port Connect Device at FXO port

Select VOICE port (1st priority) VOICE port Select FAX port (2

priority)

From the highest port number downward

Select FAX port (1st priority) FAX port Select VOICE port (2

priority)

From the highest port number downward

For the setting of "Connect Device", please refer to

9.11 CHANNEL

8.6 Call Waiting

Call waiting function for a FXS port to answer two SIP calls. When D answer a SIP call from other SIP phone or gateway, such as A. In normal condition, another incoming call dial to D will be busy, such as B to D. With Call Waiting function, the phone call dials from B to D will not be busy. Here is the possible situation.

 D keeps talking with A and hears Call Waiting Tone if B calls D.  B hears normal ring back tone without sense any different.  If D keep talking with A and ignore the Call Waiting Tone for more than 30 seconds, Call

Waiting Tone stop and the phone call return to normal condition

 If D keep talking with A and ignore the Call Waiting Tone for more than 30 seconds, B keep

hearing ring back tone for 30 seconds and listen busy tone finally.

 D can talk to B if D presses Flash button when hearing the Call Waiting Tone. Phone A is silent

when D talk to B.

 D can talk to A or to B by keep pressing Flash button to switch the two side.

DCC NO. 91710015011

 C will hear busy tone when C call to D if there is one line in call waiting status for A.

Configuration

Enable the Call Waiting function of the FXS port (D) of M3700 gateway. This function can be configured for each FXS port individually.

Web Folder: Channel\

Connection Type

A: FXS port of M3700 Series B, C: SIP Device (M3700 Series, other brand SIP gateway. SIP phone...), Normal PSTN phone call (special condition is described below)

Call waiting function works only on SIP call. So PSTN call works when it is transited as SIP call. If Inbound transit call is configured on M3700 (please refer to

7.9 Make Inbound Transit Call), then Call Waiting function is available when user dials the SIP number of this M3700 gateway itself. If no inbound transit call function is configured, it is impossible to do call waiting function.

D E

3702B

3702A SIP Phone

SIP GW

M3700 user manual

8.7 Target the Media (RTP)

For the SIP call passing through NAT, it is possible that the media would not deliver properly; owing to the RTP contact information (IP address, port number) is different from original RTP packet. This function selects different contact information for M3700 to send RTP Packets to other SIP device within far-end NAT. It designates whether to use the source contact information from the UDP/IP header (Symmetric RTP) or the contact information specified within the packet (SDP) when the unit send RTP packet

Web Folder：ADVANCED\SIP COMMON, Default Value is SDP

Example 1: Via Symmetric RTP The source contact information (IP, port number) of RTP packet is IP: 61.222.217.30, port number: 10000, but the SDP in the packet is IP: 10.13.6.18, port: 4000. In this case, please Use

Symmetric RTP

61.222.217.30 port: 10000

Network

SDP in Packet

10.13.6.18 port: 4000

M3700 (192.72.83.23, port: 10000)

M3700 tries the contact information from SDP first (IP:10.13.6.18, port number: 4000). If M3700 finds that the contact information from SDP is different from the source contact information, then it will try the source contact information, as the example above, use IP:61.222.217.30, port number:10000. It makes SIP call successful.

Example 2: Via SDP (Default) This selection ignores the source contact information (IP, port number) which M3700 received. It always sends the RTP packet to the contact information (IP, port number) described in the packet (SDP) received.

SDP in Packet

10.13.6.18 port: 4000

Network

M3700 (192.72.83.23, port: 10000)

Send RTP to

10.13.6.18 port: 4000

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9. WEB MANAGEMENT INTERFACE

The Tree Architecture of Web Management is shown below

HOME BASIC GENERAL

IP SETTING

GENERAL SIP COMMON SIP OUTBOUND AUTHENTICATION SIP INBOUND ATHENTICATION STUN DIALING PLAN FILE TEMPLA TE

ADVANCED

INBOUND TRANSIT (for gateway has FXO port. Gateway without

FXO port does not have this page) CHANNEL PHONE BOOK

ACCESS CODE

M3700 user manual

9.1 BASIC / GENERAL

Category Section Description Default Setting

Region ID Display region ID.(Read only) 0 Software Version

Display software version.(Read only)

BootRom Version

Display BootRom Version.(Read only)

Hardware Version

Display hardware Version.(Read only)

Card Type Display card type. (Read only) Up-Time Display the use time since from system

reboot.(Read only)

Information

MAC Address

Display MAC address.(Read only)

Date Show the date Time Show the time

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Time Source

Select the time server to synchronize the time of this gateway ♦ Registrar: Get the time data from the

Registrar Server.

♦ NTP Server: Get the time data from

the NTP Server

Registrar

NTP Server Input the address if the system use

NTP server as time synchronization source. The gateway will synchronize with the NTP Server once a day. If the NTP server inputted here is not available or fail to response, the gateway will retry it every 5 minutes. The gateway has its own clock, so the clock will keep going according to last synchronization time. For NTP server information, please refer to http://www.ntp.org

Time Zone Select local system time zone. Select

correct Time Zone.

Time Configuration

Daylight saving

ON: Enable daylight saving. OFF: Disable daylight saving.

OFF

Signaling Port

UDP port to transfer signal packets. It can be setting in the range of 0 to

65535. (Must reboot system to apply changes)(Only support VuTek device)

protocol

RTP Base Port

Base of UDP port to receive RTP packets. It can be setting in the range of 0 to 65534.( Must be Even, after setting this item, please reboot system to apply changes)

4000

System Restart

Restart Mode

None: Not to restart system. Cold restart: Cold restart. Warm restart: Warm restart.

None

M3700 user manual

9.2 IP SETTING

Category Section Description Default Setting

IP State The way to obtain IP address:

Manual: Entered by user (Static IP) Auto(DHCP): Assigned by DHCP server PPPoE: Assigned by PPPoE of ISP

Manual

IP Settings

Current Setting Display the configured IP

address, subnet mask address and default gateway. (Read only)

192.168.0.2

255.255.255.0

192.168.0.1

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Change To Enter the IP address that will

be used after next restart, Including: IP Address Subnet Mask Address Default Gateway (This item is used only on Manual mode of IP Setting.)

Account The user’s account of PPPoE

protocol, provided by ISP.

Password The user’s password of PPPoE

protocol. Confirm Password

Confirm the user’s password of

PPPoE protocol.

PPPoE Settings

Service Name The service name of PPPoE

account, provided by ISP.

(Most ISP doesn’t need this) Primary Address The primary address of DNS

server. The default setting

would be different according to

the local area. In Taiwan, the

default setting is 168.95.1.1.

168.95.1.1

DNS Server

Secondary Address

The secondary address of

DNS server. User Name The user’s name of Web

Management Interface.(12

character)

WEB

Password The password of Web

Management Interface.( 6

character)

Web Password

Password Confirm

Enter the password again to

confirm it.

M3700 user manual

9.3 ADVANCED / GENERAL

Category Section

Description

Default Setting

Flash Button Flash Time System confirmed

“Flash” time.

200 msec

Duration The duration to send a

DTMF.

100 msec Touch Tone (DTMF)

Inter-digit The inter-digit time of

sending string of DTMF digits.

100 msec

Guard Time Line The time defines how

long the system will not accept incoming call after previous call has been disconnected.

0.8 sec

Dial Ending Time Dial Ending The time specifies how 4

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Time long to end the dialing

number if a ‘#’ digit is missing.

1-10 (seconds)

T.38 Fax Relay Redundancy Number of times to retry

T.38 Fax protocol. Use more Redundant packet when network is unstable. No Redundant packet 1 Redundant packet 2 Redundant packets 3 Redundant packets 4 Redundant packets

Voice Jitter Buffer

Select the buffer time to suppress voice delay vibration

1. Auto, the system detects it automatically.

2. Other selection from 20ms~460ms

Auto

Frequency f1, f2

(300 ~ 3000Hz)

Busy Tone Spec

Cadence on, off. The on and off

duration in playing the tone

(100 ~ 5000ms)

Frequency f1, f2

(300 ~ 3000Hz)

Reorder Tone Spec

Cadence on, off. The on and off

duration in playing the tone

(100 ~ 5000ms)

M3700 user manual

9.4 SIP COMMON

Section Item Field Description Default Port and Header Port The control port number of SIP protocol. 5060

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Section Item Field Description Default

Header Form

Select ‘Standard’ or ‘Compact’ to be the header format of SIP packet. When Compact is selected, the header will be shorter and it saves bandwidth.

Standard

Domain Name

Domain name or IP address of proxy. Empty

Disable

Outbound Proxy Setting

Port Control port number of SIP protocol. 5060

Registrar Setting Domain

Name

Domain name or IP address of proxy that you want to register.

Empty Disable

Out-band DTMF Control Control Out-of-Band DTMF

Enable/Disable

Disable

Incoming Call Screening

Screening Disable: Accept all incoming SIP call

Enable: This gateway only accepts

incoming call through SIP Proxy.

Disable

NAT Signalling Keep Alive

Control Port number mapping may change if the

connection to pass through some NAT device is timeout. This function sends Dummy Packet to Proxy server every 50 seconds to keep the port number via NAT intact. Disable: Does not send Dummy Packet Enable: Send Dummy Packet

Disable

Target the media (RTP)

Via Select the contact information (IP

Address, Port Number) to pass through NAT device. Please refer to

8.7 Target the Media SDP: via SDP Symmetric RTP: via Symmetric RTP

SDP

M3700 keeps to register the SIP proxy before expire time to insure registration

3600

Codecs Selection Codec

Type

G.729AB: Mark the selection to Enable

G.729AB Codec

Enable

M3700 user manual

Section Item Field Description Default

G.723.1: Mark the selection to Enable

G.723.1 Codec

Enable

PCMU: Mark the selection to Enable

PCMU Codec (G.711 u Law)

Enable

PCMA: Mark the selection to Enable

PCMA Codec (G.711 A Law)

Enable

Codec

Priority

You can select the codec priority for your requirement.

G729-G723-P CMU-PCMA

SIP Entity Control (2 buttons) Register All: Let all SIP Entities of this

machine to register the SIP Proxy De-Register All: Let all SIP Entities of this machine to de-register (quit) from the SIP Proxy

None

SIP Entity Select an entity and click Select button

to display follow items’ setting of SIP entity section. Select: Select Button Register: Register Button De-Register: Cancel Register Button

Entity Control

Select Enable/Disable Enable

Show the register status, if it shows Registered means successful. (Read only) Register: Register Button De-Register: Cancel Register Button

Empty

SIP Entity

CLIR Calling Line Identification Restriction

Disable: Send caller ID to SIP proxy when user make SIP call Enable: Don’t send caller ID when user make SIP call. Note that for some SIP Proxy Server, the SIP call is failed if no caller ID is sent. Please set “CLIR” Disable for this case. That’s the reason why default value is disable.

Disable

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Section Item Field Description Default

Address Enter SIP phone number of the port.

The phone number general assigned by SIP service provider.

Empty

Public Address Setting

Default Account

Account information for registering SIP Proxy Username: It may the same as your SIP number Password: Password for Authentication Confirm Password: Reconfirm Password

Contact Address Setting

Current Setting

Display current setting of Contact Address. It will be the same as the Username of Public Address Setting at this page of web if that field is configured

(Read Only)

RFC 2833 DTMF 2833

DTMF

Enable: Enable RFC 2833 DTMF. Negotiate: Encode DTMF to message and decode it back at destination. Never: Convert DTMF to voice and sent by RTP packets.

Negotiate

Forward Address

Enter a SIP account (Public Address) forward. When users dial into the SIP Entity, the call will be forwarded to the number. Both SIP calls and FXO called can be forwarded.

Empty

Forward To

Type N/A: All incoming calls are forward.

Busy: When the SIP entity is busy, the calls will be forward. No Answer: When the SIP entity is no answer about 30 seconds, the calls will be forwarded.

N/A

SIP Entity Members

Channel Show the all channels Depend on

gateways

M3700 user manual

Section Item Field Description Default

Entity Show ‘+ ‘ means the SIP entity is for the

channel.

Empty

9.5 SIP OUTBOUND AUTHENTICATION

Section Item Field Description Default

Maximum Maximum number of entries

allowed

(Read Only) 50

Entered Number of entries of

authentication entered.

(Read Only) 0

List of entries (Read Only)

SIP Outbound Authentication

Entries List

Entity: Which entity that you select. Realm: Domain name or IP address. Username: Username of authentication.

The gateway creates default entry according to the Public Address Setting for easy registration. Please refer to

7.3 SIP Entity and 7.4 SIP Outbound Authentication

Empty

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Section Item Field Description Default

Update Entry

Enter the information of outbound authentication Entity: Select an entity. Realm: Domain name or IP address. Username: Enter Username of authentication. Password: Enter password of authentication. Confirm Password: Enter password again for confirmation.

Empty

Delete Entry

Delete the information of outbound authentication Entity: Select an entity. Realm: Domain name or IP address.

Empty

9.6 SIP INBOUND ANTHENTICATION

Section Item Field Description Default SIP Inbound Authentication

Realm Enter domain name, IP address or word

string.

Empty

M3700 user manual

Section Item Field Description Default

Maximum Maximum number of

entries allowed

(Read Only) 20

Entered Number of entries of

authentication entered.

(Read Only) 0

Display the entries (Read Only) Entries List Entity: Which entity that you select. Username: Username of authentication.

Empty

Update Entry Enter entries of authentication

Entity: Which entity that you select. Username: Username of authentication. Password: Password of authentication. Confirm Password: Enter password again for confirmation.

Empty

Delete Entry Delete entries of authentication

Entity: Which entity that you want to delete. Username: Username of authentication.

Empty

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9.7 DIALING PLAN

Section Item Field Description Default Dialing Method Control Dialing Plan: Use dialing plan rule

Transparent: All number dialed is passed to Proxy Server Transparent (with Digitmap): All number dialed is passed to Proxy Server with digit map control For the detail of this function, please refer to

8.1 Dialing Plan

Dialing Plan

DIALING PLAN Maximum Maximum number of

entries allowed

(Read Only) 100

M3700 user manual

Section Item Field Description Default

Entered Number of entries of

authentication entered.

(Read Only) 1

Display the entries (Read Only) List The default value “x“ means that all numbers that you dial will first go through SIP proxy.

Add Dialing Plan Enter numbers. Example: 050. Empty

Delete Entry Enter numbers for delete. Empty Dial In Rewriting Rule

Control Digits dialed from M3700 can be

rewrite to different digits and sent to SIP Proxy. Enable/Disable

Disable

Capacity The max set of rewrite number List List the entries of original digits

and the rewrite digits Pattern: the pattern that user may

dial

Rewrite: the converted number if

user dials the same digit in pattern column.

Add Dialin (button) Pattern: Add the pattern that user

may dial

Rewrite: Add the converted

number if user dials the same

digit in pattern column. Fill in digits and click the Add Dialin button

Del Dialin (button) Fill in the Pattern digit that will be

deleted and click Del Dialin button

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9.8 FILE TEMPLATE

File template is the text data of your customized configuration. You can keep it for backup purpose. Configure other M3700 with this text file can save time to re-configure it.

For this purpose, copy all text in this page (does not include left title column), paste to Windows Notepad and save it as SIP33XX.MEM (for example, SIP3304.MEM. You can connect to gateway by FTP to see the file name of your gateway)

M3700 user manual

This content of this file can be modified, then upload the file to Gateway. If there is lots of data need to create or modify, use this way can save lots of time.

For the procedure to use FTP client to connect this gateway, please refer to

11.2 Software Update

Note: After you had uploaded MEM file back to gateway, for those setting that need no to restart the machine, it will take effect immediately; for those setting that need to restart the machine, you have to restart the machine to take effect.

9.8.1 Template of MEM file

Format: Parameter = value (description) You can refer to the text file below as example.

[SIP-COMM] SIP-Port = 5060 Header-Form = 0 # (0/1, Standard/Compact) Out-Proxy-Domain = "outboundproxy.com" Out-Proxy-Status = 1 # (0/1, Disable/Enable) Out-Proxy-Port = 5060 Registrar-Domain = "registrar.com" Registrar-Status = 1 # (0/1, Disable/Enable) Out-Of-Band-DTMF = 0 # (0/1, Disable/Enable) Incoming-Call-Screen = 0 # (0/1, Disable/Enable) NAT-Keep-Alive = 0 # (0/1, Disable/Enable) Target-The-Media = 0 # (0/1, SDP/Symmetric RTP)

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Codecs-Select ion = "1111" # (G729:1000,G723:100,G711U:10,G711A:1) Codec-Priority = 0 #(0~23) (refer to webpage) Hotline-Control = 0 # (0/1, Disable/Enable) RTP-Base-Port = 10000 Time-Source = 1 # (0/1, Registrar/NTP Server) NTP-Server = "ntpserver.com" Time-Zone = 24 # (0~29) (refer to webpage) DayLight-Saving = 0 # (0/1, Off/On)

[SIP-ENTITY] Entity-No = 1 Entity-Control = 1 # (0/1, Disable/Enable) CLIR = 0 # (0/1, Disable/Enable) Public-Address = "user@registrar.com" Default-Account-User = "username" Default-Account-PASS = "password" RFC-2833-DTMF = 1 # (0/1, Never/Negotiate) Forward-Address = "user2@registrar.com" Forward-Type = 0 # (0/1/2/3, None/All/Busy/No Answer)

[CHANNEL-CONFIG] Channel-No = 1 Join-SIP-Entity = 1 # (0 for None) Control = 1 # (0/1/2, IN_Only/BothWay/Disable) DND = 0 # (0/1, Disable/Enable) Slience-Suppress = 0 # (0/1, Disable/Enable) Connect-Device = 0 # (0/1, Phone/Fax) Battery-Reverse = 0 # (0/1, Off/On) Auto-Answer = 1 # (0/1/2, Disable/Enable/Enable w/Pincode) Call-Waiting = 1 # (0/1, Disable/Enable) T38-Fax = 1 # (0/1, No/Yes) Voice-Input-Gain = 1 # (0~12, -6~6) Voice-Output-Gain = 1 # (0~12, -6~6)

[SIP-OUTBOUND-AUTH] # format: (entity realm username password), entity 0 for all 0 "realmA" "realmA_user" "password" 1 "realmB" "realmB_user" "password"

M3700 user manual

[SIP-INBOUND-INFO] Realm = "realm.com"

[SIP-INBOUND-AUTH] # format: (entity username password), entity 0 for all 0 "username_for_all" "password" 1 "username_for_1" "password"

[STUN] STUN-Server-Control = 0 # (0/1, Disable/Enable) NAT-WAN-IP = "223.223.223.223" STUN-Refresh-Time = 60 # unit: seconds

[STUN-SERVER] # format: (ip_address port)

223.223.223.224 3478

[SIP-PHONE-BOOK] # format: (index user_part host_part port via_proxy) 8888 "user" "registrar.com" 5060 0 # Via Proxy(0/1, No/Yes)

[TELEPHONY] DIAL-END-TIME = 1 # (1~10) (refer to webpage) T38-RELAY = 0 # (0~4) (refer to webpage)

9.8.2 Related Configuration at Web Page

Each Text parameter can be refered to a setting in web page. The table below shows the relation. For the configuration of each setting, please refer to related chapter of this manual.

Parameter Description and Web Folder [SIP-COMM]

Configuration of [SIP-COMM]

SIP-Port = 5060 Web Path: ADVANCED\SIP COMMON

Header-Form = 0 # (0/1, St andard/Comp act) W eb Path: ADVANCED\SIP COMMON

Out-Proxy-Domain = "sip99.yip.com" Web Path: ADVANCED\SIP COMMON

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Out-Proxy-Status = 1 # (0/1, Disable/Enable)

Web Path: ADVANCED\SIP COMMON

Out-Proxy-Port = 5060 Web Path: ADVANCED\SIP COMMON

Registrar-Domain = "sip99.yip.com" Web Path: ADVANCED\SIP COMMON

Registrar-Status = 1 # (0/1, Disable/Enable) Web Path: ADVANCED\SIP COMMON

Out-Of-Band-DTMF = 0 # (0/1, Disable/Enable)

Web Path: ADVANCED\SIP COMMON

Incoming-Call-Screen = 0 # (0/1, Disable/Enable)

Web Path: ADVANCED\SIP COMMON

NAT-Keep-Alive = 0 # (0/1, Disable/Enable) Web Path: ADVANCED\SIP COMMON

Target-The-Media = 0 # (0/1, SDP/Symmetric RTP)

Web Path: ADVANCED\SIP COMMON

Codecs-Select ion = "1111" # (G729:1000,G723:100,G711U:10,G711A:1)

Web Path: ADVANCED\SIP COMMON Set to "0" if that codec is not used.

Codec-Priority = 0 #(0~23) (refer to webpage)

Web Path: ADVANCED\SIP COMMON

M3700 user manual

For the sequence of selection, please refer to web page.

Hotline-Control = 0 # (0/1, Disable/Enable) Web Path: PHONEBOOK

RTP-Base-Port = 4000 Web Path: BASIC\GENERAL

Time-Source = 1 # (0/1, Registrar/NTP Server)

Web Path: BASIC\GENERAL

NTP-Server = "ntpserver.com" Web Path: BASIC\GENERAL

Time-Zone = 24 # (0~29) (refer to webpage) Web Path: BASIC\GENERAL

For the sequence of selection, please refer to web page.

DayLight-Saving = 0 # (0/1, Off/On) Web Path: BASIC\GENERAL

[SIP-ENTITY] Configuration of SIP Entity Entity-No = 1 Web Path: ADVANCED\SIP COMMON

Entity-Control = 1 # (0/1, Disable/Enable) Web Path: ADVANCED\SIP COMMON

CLIR = 0 # (0/1, Disable/Enable) Web Path: ADVANCED\SIP COMMON

Public-Address = "user@registrar.com" Web Path: ADVANCED\SIP COMMON

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Default-Account-User = "username" Web Path: ADVANCED\SIP COMMON

Default-Account-PASS = "password" Web Path: ADVANCED\SIP COMMON

RFC-2833-DTMF = 1 # (0/1, Never/Negotiate)

Web Path: ADVANCED\SIP COMMON

Forward-Address = "user2@registrar.com" Web Path: ADVANCED\SIP COMMON

Forward-Type = 0 # (0/1/2/3, None/All/Busy/No Answer)

Web Path: ADVANCED\SIP COMMON

[CHANNEL-CONFIG] Configuration of Channel Channel-No = 1 Web Path: Channel

Join-SIP-Entity = 1 # (0 for None) Web Path: Channel

Control = 1 # (0/1/2, IN_Only/BothWay/Disable)

Web Path: Channel

DND = 0 # (0/1, Disable/Enable) Web Path: Channel

Slience-Suppress = 1 # (0/1, Disable/Enable)

Web Path: Channel

Connect-Device = 0 # (0/1, Phone/Fax) Web Path: Channel

Battery-Reverse = 0 # (0/1, Off/On) Web Path: Channel

M3700 user manual

Auto-Answer = 0 # (0/1/2, Disable/Enable/Enable w/Pincode)

Web Path: Channel

Call-Waiting = 0 # (0/1, Disable/Enable) Web Path: Channel

T38-Fax = 1 # (0/1, No/Yes) Web Path: Channel

Voice-Input-Gain = 0 # (0~12, -6~6) Web Path: Channel

Voice-Output-Gain = 0 # (0~12, -6~6) Web Path: Channel

[SIP-OUTBOUND-AUTH] # format: (entity realm username password), entity 0 for all

Configuration of SIP Outbound Authentication

1 "ABC" "Lester" "1234" Web Path: ADVANCED\SIP OUTBOUND

AUTHENTICATION

0 "DEF" "Jack" "5678" Web Path: ADVANCED\SIP OUTBOUND

AUTHENTICATION

[SIP-INBOUND-INFO] Configuration of SIP Inbound Authentication Realm = "realm.com" Web Path: ADVANCED\SIP INBOUND

AUTHENTICATION

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[SIP-INBOUND-AUTH] # format: (entity username password), entity 0 for all

Configuration of SIP Inbound Authentication

1 "Lester" "1234" Web Path: ADVANCED\SIP INBOUND

AUTHENTICATION

0 "Jack" "5678" Web Path: ADVANCED\SIP INBOUND

AUTHENTICATION

[STUN] Configuration of STUN STUN-Server-Control = 0 # (0/1, Disable/Enable)

Web Path: ADVANCED\STUN

NAT-WAN-IP = "0.0.0.0" Web Path: ADVANCED\STUN

STUN-Refresh-Time = 30 # unit: seconds Web Path: ADVANCED\STUN

[STUN-SERVER] # format: (ip_address port)

Configuration of STUN Server

61.222.217.99 3479 Web Path: ADVANCED\STUN

[SIP-PHONE-BOOK] # format: (index user_part host_part port via_proxy)

Configuration of SIP Phone Book

8888 "Lester" "registrar.com" 5060 0 # Via Web Folder: PHONEBOOK

M3700 user manual

Proxy(0/1, No/Yes)

[TELEPHONY] Configuration of Telephony DIAL-END-TIME = 4 # (1~10) (refer to webpage)

WEB Folder: ADVANCED\GENERAL

T38-RELAY = 3 # (0~4) (refer to webpage) WEB Folder: ADVANCED\GENERAL

9.9 INBOUND TRANSIT

Only FONEM3700 gateway with FXO port has this web page.

DCC NO. 91710015011

Group Field Description Default Value

Warning Time This gateway will send warning tone periodically to

check if the line is still alive. If calling side fail to press any key after hearing the warning tone, the line will be disconnected.

Transit call

Release Call by Checking RTP

This gateway will check the RTP packet periodically to verify if the line is still alive. If no RTP packet is found, the gateway will disconnect the call. When this value is set to "0", means the gateway will not check the RTP packet

Maximum Display no. of password can

be accepted

(Read only) 32

Entered Display the no. of password

had been entered

(Read only) 0

Entries List List the detail data of password

had been entered

(Display) Only) Blank

Add Passwords Enter a new password, any combination of digits

(0~9), less than 9 characters. The password will be used at Pincode for auto answer function

Blank

Password For Inbound Transit

Delete Passwords

Enter the password to be deleted, refer the detail

data under Entries List

Blank

M3700 user manual

9.10 STUN

Section Item Field Description Default STUN Server Control Enable or Disable STUN Server service.

Disable

NA T W AN IP Address Input this NAT WAN IP helps you to pass

through NA T without using STUN server. The port number inside and outside NAT should be the same. NAT WAN IP is the fixed Public IP that used on NAT device Note: If you disable STUN server and input NAT WAN IP here, the RTP (normally 4000) and Signaling (normally

5060) port number inside and outside NAT must be the consistent, and Server Port need to be configured on NAT device.

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Section Item Field Description Default

Maximum Maximum number of

entries allowed

(Read Only) 5

Entered Number of entries of

STUN server that have been entered.

(Read Only) 0

List Display all of servers that

have been entered.

(Read Only)

Add Add a stun server

IP Address: Enter IP address or Domain Name Port: Enter port number of service.

Empty

STUN Server Setting

Delete Delete a stun server

IP Address: Enter IP address. Port: Enter port number of service.

Empty

NAT Type Type Display NAT type (Read Only) Unknown STUN Refresh Time Interval It defines how long the device will send

a binding request packet with discard flag on to STUN server.

Mapping List List

My ip/port: shows the private IP and port number. Global ip/port: Display public IP and port number.

(Read Only) Empty

M3700 user manual

9.11 CHANNEL

Category Section Description Default

Setting Channel Channel number: 1 Information Channel

Type

Display port type. (Read only) Phone: FXS Interface, connect to telephone set or Fax machine. Line: FXO Interface, connect to phone line.

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Channel Control

For FXS port: Bothway: Can make and accept IP call and PSTN call from this channel Disable Disable all functions of this port.

For FXO port: IN_Only: Accept calls from PSTN only Bothway: Accept call from PSTN or call dial from FXS Disable: Disable all functions of this port.

Enable

Current State Display the current state of this

port. (Read only)

Enable/ Disable. Do not Disturb

Enable/Disable does not

disturb function

Disable

Silence Suppression

Enable/Disable the function. Enable

2833 In use Yes

(Read only)

Join SIP Entity

Select an Entity for SIP.

Both FXS and FXO ports can

join SIP Entity

Connect Device

Phone: Connect to this port is

regular phone

FAX: Connect to this port is

FAX machine. Codec will be

fixed on G.711 if SIP-based

T.38 codec negotiation fails.

Both FXS and FXO ports can

select their Connect Device

Phone

M3700 user manual

Battery Reverse

This mechanism will reverse

the polarity promptly that help

some PBX to identify the start

and end of each call

ON: Enable the function

OFF: Disable the function

OFF

Auto Answer This unit auto answer the call

from FXO

Disable: Disable Auto Answer

Enable: Enable Auto Answer

Enable w/ Pincode: Enable

Auto Answer and Pincode

verification.

Forward to SIP: When

incoming FXO calls is answer,

it is forwarded to a SIP

destination address. (For this

SIP address, please refer to

the configuration on Web

Folder: ADV ANCED\SIP

COMMON\ Forward To field

and it depend on which SIP

Entity it joins)

Disable

Call Waiting Call waiting function for

answering two incoming SIP

VoIP phone calls

Enable: Enable call waiting

Disable: Disable call waiting

Disable

T.38 FAX Relay Control Yes: Use T.38 as FXS protocol

No: Don't use T.38 as FAX

protocol. If user send or receive

FAX by this port, gateway can

use G.711 (PCMU, PCMA) to

pass-through FAX, please refer

8.4 FAX

Input Gain Adjust Voice input Gain 0 Voice Output Gain Adjust Voice output Gain 0

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9.12 PHONE BOOK

Section Item Field Description Default Apply to Hotline Control Enable or Disable the hotline function to

M4600 SIP Line or other SIP device to make hotline call.

Disable

Maximum Maximum number of entries

allowed

(Read Only) 200

Entered Number of entries of phone

books entered.

(Read Only) 0

Display phone books (Read Only) Entries

List

Index: Dialing number Public Address: SIP account. Port: Port number. Via Proxy: Via proxy or not.

Empty

Update Entry

Enter entries Index: Enter dialing number Public Address: Enter SIP account. Port: Enter port number Via Proxy: Select via Proxy or not

Empty

SIP Phone Book

Delete Entry

Delete entries Index: Enter the index for delete.

Empty

M3700 user manual

10. Use Private IP (Behind NAT)

Using a Private IP in a NAT Environment The unit is able to communicate with other units under a NAT environment using Private IP addresses on the LAN side of your IP Sharing device. However you must configure the IP Sharing device to treat the unit as a Virtual Server using UDP port 5060,2000.

You will have to ask MIS personnel to enable the ports listed in the following table.

Packet Modes Using Ports SIP Signal Packets UDP 5060 Signaling Port UDP 2000 RTP Base Port UDP 4000 FTP software upgrade TCP 21 Web management TCP 80

If you want to use private IP behind NAT and Proxy Server is in Internet, you must need to enable STUN service or configure NAT WAN IP. Please refer to

9.10 STUN. If the system is installed in

VPN, it is not necessary to Enable Stun.

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11. File Management

11.1 File Types

The naming convention to the file type of FONEM4496 is listed in the following table:

File Name File Type Description

SIP3302.CFG SIP3304.CFG

SIP33XX.CFG

System

configuration file

File of system configuration

SIP3302.RUN SIP3304.RUN

SIP33XX.RUN

Executing file System Software

SIP3302.WEB SIP3304.WEB

SIP33xx.WEB

Web file Page for web browser

SIP3302.MEM SIP3304.MEM SIP33xx.MEM

Text file

MEM setting file can be downloaded by FTP to PC; open file and modify the contents using NOTEPAD or other word processing tool; then uploaded the file to system.

11.2 Software Update

11.2.1 Software update via FTP

Preparation before Updating FIRMWARE

1. Power on the Conference Bridge

2. Get Windows based PC ready

3. LAN cable is well connected (for FTP)

4. Configure the IP, Subnet, and Default Gateway of this gateway and PC

5. Get the file of update “GW FIRMWARE” ready

M3700 user manual

Software Update by FTP for File Type RUN and WEB

1. Execute FTP Client Software, e.g. CuteFTP Enter IP Address, User Name (default is FTP), Password (the password of FTP and Console is same, and the default is blank), and the Port Number to 21

2. Click button Connect to get connection between gateway and FTP Client. The files of the gateway will be displayed on the window if the connection is successful.

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3. Select the file with extension of .RUN and click button Upload and then Yes to overwrite. (Please notice that the file name must be same as the file name in the Gateway, e.g. SIP3304

.RUN).

4. After the file is overwritten (you may check if the time of the file is updated), Gateway has to run Cold Start to store the configure file, then the updating is effective.

5. Select the file with extension of .WEB and click button Upload (Please notice that the file name must be same as the file name in the Gateway, e.g. SIP3302

.WEB). And repeat the

step 3 ~ 4.

6. Check if the uploading is successful, you enter the Web Management Page to examine the version of software. (Web Folder: BASIC\GENERAL)

Check if the version is correct

M3700 user manual

12. Appendix

12.1 Appendix A: Phone-Set Command

Pick up the handset and listen for the dialing tone. Dial “##0000 and listen for three consecutive tones before setting the following parameters. After input the parameters, please dial ‘# to end the configuration. Note: If Dialing Mode is configured to Transparent, (refer to section

8.1 Dialing Plan) then all digits

you dial is passed to SIP Proxy and all Phone-Set Command is disabled.

Command Description

Parameters

01 IP State 0 : static; 1: DHCP; 2: PPPoE 02 IP Address xxx*xxx*xxx*xxx 03 Subnet Mask xxx*xxx*xxx*xxx 04 Default Gateway xxx*xxx*xxx*xxx 05 Primary DNS Server

xxx*xxx*xxx*xxx

06 Second DNS Server

xxx*xxx*xxx*xxx

07 Select Signaling Port 0~65535 08 Select RTP Base Port 0~65534 (limit to even port number only) 09 PPPoE username User name (use the mapping table to map

character into digits)

10 PPPoE password Password (use the mapping table to map

character into digits)

11 DND Do not Disturb, this line accept dial out call only.

All incoming call is terminated. 0 : Disable ; 1:

Enable 12 SIP Forward State 0 : Disable ; 1: Enable; 2: Busy; 3: No Answer 13 SIP Forward To

Number

The SIP number that this line will forward to. The

Forward To address is "key in phone-set

number@SIP proxy registered". For example,

73796@fwd.pulver.com, 73796 is the number

you key-in by phone-set. fwd.pulver.com is the

registered proxy of this gateway. 14 Change Service Port 1:FTP; 2:HTTP 3:Telnet (Port: 0-65535)

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15 Change WEB

Password

6 digits

16 Change FTP

Password

6 digits

17 Register or

De-register (quit) the SIP Entity registration

0: De-Register; 1: Register

40 Listen for the IP

Address

(ending ”#” is not required)

41 Listen for the Subnet

Mask

(ending ”#” is not required)

42 Listen for the Default

Gateway

(ending ”#” is not required)

46 Listen for WEB, FTP,

Telnet Port

1:FTP; 2:HTTP 3:Telnet

47 Listen for Current

Public Address

(ending ”#” is not required)

95 Region ID 2 digits 97 Reset unit to Factory

Default values

1: reset all; 2: keep IP; 3: region specific

98 System Warm Restart 1: do it

M3700 user manual

12.2 Appendix B: Console Command

User Exec commands

Enable Turn on privileged commands

Exit Exit from the EXEC Help Description of the interactive help system Show Show running system information

show

Dns Show the IP address of domain name server

ethernet FastEthernet port status and configuration history Display the session command history Ip Display IP configuration running-config Show current operating configuration version System hardware and software status

Privileged Mode

Configure Enter configuration mode

Delete Reset configuration Disable Turn off privileged commands Exit Exit from the EXEC Help Description of the interactive help system Ping Send echo request to destination Probe-hook probe busytone cadence Probe-remove stop probe busytone cadence Reload Halt and perform cold start Restart Halt and perform warm start Show Show running system information

Global Mode

Dbflush DataBase flush

Dns Set the IP address of domain name server End Exit from configure mode to privileged mode Exit Exit from configure mode Help Description of the interactive help system Ip Global IP configuration subcommands Log Control log output No Negate a command or set its defaults pppoe PPPoE configuration subcommands regional_id Set regional id service_port Set service port number

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12.3 Specifications

Voice Interface

FXS interface

Loop start, 2 wire Feeding Voltage: 20V Feeding Current: 30 mA

FXO interface Loop start, 2 wire

Connectors

RJ-11 Connectors (3702/3704)

IDC Connectors (3708/3716) Voice compression G.711/G.723/G.729AB Silence suppression VAD, CNG Echo cancellation G.165/G.168 16ms Jitter buffer Adaptive jitter buffer management Gain control In/Out +/-6db Transport protocols RTP, RTCP Call control protocol Pure SIP Network Interface

Number of ports

Two Ethernet ports and One console port (for 4,8,16

ports models) Interface 10BASE-T/100BASE-TX Auto-negotiation Connectors RJ-45 Connectors General Spec

Dimension

M3702: 190mm x 110mm x 25 mm

M3704: 172mm x 177mm x 35 mm

M3708: 440mm x 44mm x 254 mm

M3716: 440mm x 66mm x 254 mm Power Voltage: 100-240 VAC, Frequency: 50/60 Hz

Power consumption

M3702: 8 W

M3704: 12W

M3708/3716: 70W

Working environment

Operating temperature: 0 to 50℃

Storage temperature: -10 to 70℃ EMI FCC part 15 Class B . CE Mark PTT FCC part 68 , NALTE , iDA , JATE Safety cUL , CCIB , CB

M3700 user manual

12.4 Mapping table of characters used in PPPoE

Character Digits to key -in Character Digits to key-in

0 30 X 58 1 31 Y 59 2 32 Z 5*0 3 33 a 61 4 34 b 62 5 35 c 63 6 36 d 64 7 37 e 65 8 38 f 66 9 39 g 67

@ 40 h 68

A 41 i 69 B 42 j 6*0 C 43 k 6*1 D 44 l 6*2 E 45 m 6*3 F 46 n 6*4 G 47 o 6*5 H 48 p 70

I 49 q 71

J 4*0 r 72

K 4*1 s 73

L 4*2 t 74 M 4*3 u 75 N 4*4 u 76 O 4*5 w 77 P 50 x 78 Q 51 y 79 R 52 z 7*0 S 53 = 3*3 T 54 . 2*4 U 55

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V 56

W 57

12.5 Region ID

Country Region ID Country Region ID Country Region ID

Argentina 01 France 12 Singapore 36

Australia 02 Germany 13 Slovenia 38

Philippines 03 Hong Kong 15 South Africa 39

Portugal 04 India 18 Spain 40

Brazil 05 Italy 22 Switzerland 42

Canada 06 Japan 23 Taiwan 43

China 07 Korea 24 Thailand 44

Russia 08 Malaysia 26 British 46

Sweden 09 Mexico 27 USA 47

Vietnam 10 Netherlands 28

Belgium 11 New Zealand 29

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

Specifications and Main Features

Frequently Asked Questions

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