VoxStack Series Wireless Gateways User Manual

OpenVox Communication Co Ltd

VoxStack Series Wireless Gateways

User Manual
 VoxStack Series Wireless Gateways User Manual

OpenVox Communication Co.,Ltd

Address: Room 624, 6/F, Tsinghua Information Port, Book Building, Qingxiang Road,
Longhua Street, Longhua District, Shenzhen, Guangdong, China 518109
Tel: +86-755-66630978, 82535461, 82535362
Business Contact: [email protected]
Technical Support: [email protected]
Business Hours: 09:00-18:00(GMT+8) from Monday to Friday
URL: www.openvox.cn

Thank You for Choosing OpenVox Products!

 VoxStack Series Wireless Gateways User Manual

Copyright
Copyright© 2020 OpenVox Inc. All rights reserved. No part of this document may be
reproduced without prior written permission.

Confidentiality
Information contained herein is of a highly sensitive nature and is confidential and
proprietary to OpenVox Inc. No part may be distributed, reproduced or disclosed
orally or in written form to any party other than the direct recipients without the
express written consent of OpenVox Inc.

Disclaimer
OpenVox Inc. reserves the right to modify the design, characteristics, and products at
any time without notification or obligation and shall not be held liable for any error
or damage of any kind resulting from the use of this document.
OpenVox has made every effort to ensure that the information contained in this
document is accurate and complete; however, the contents of this document are
subject to revision without notice. Please contact OpenVox to ensure you have the
latest version of this document.

Statement
This document applies to all VoxStack series gateways, including GSM/WCDMA/LTE.

Different types of gateways may have functional differences. For details, please contact

OpenVox sales or technical support.

Trademarks
All other trademarks mentioned in this document are the property of their respective
owners.
 VoxStack Series Wireless Gateways User Manual

Revision History
Version Date Detail

1.0 2019/04/04 Initial

 VoxStack Series Wireless Gateways User Manual

Contents
1. Overview .................................................................................................................... 4

1.1 What is VoxStack Series Wireless Gateways？ ................................................ 4

1.2 Application ....................................................................................................... 5

1.3 Main Features .................................................................................................. 6

1.4 Physical Information......................................................................................... 8

1.5 Login ................................................................................................................. 9

2. System ...................................................................................................................... 10

2.1 Status.............................................................................................................. 10

2.2 Time ............................................................................................................... 12

2.3 Login Settings ................................................................................................. 13

2.4 General ........................................................................................................... 14

2.5 Tools and Information .................................................................................... 15

2.6 Setting Wizard ................................................................................................ 17

3. Module ..................................................................................................................... 19

3.1 Module Settings ............................................................................................. 19

3.2 Call Forwarding .............................................................................................. 24

3.3 Call Waiting .................................................................................................... 25

3.4 DTMF .............................................................................................................. 26

3.5 BCCH............................................................................................................... 27

3.6 Toolkit ............................................................................................................. 28

3.7 Module Update .............................................................................................. 29

3.8 Call and SMS Limit .......................................................................................... 30

4. VOIP.......................................................................................................................... 36

4.1 VOIP Endpoints .............................................................................................. 36

4.1.1 Add New SIP Endpoint ........................................................................ 36

4.1.2 Add New IAX2 Endpoint ...................................................................... 44

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4.2 Batch SIP Endpoints ....................................................................................... 49

4.3 Advanced SIP Settings .................................................................................... 51

4.4 Advanced IAX2 Settings ................................................................................. 59

4.5 SIP Account Security ...................................................................................... 65

5. Routing ..................................................................................................................... 67

5.1 Call Routing Rules........................................................................................... 67

5.2 Groups ............................................................................................................ 71

5.3 Batch Creating rules ....................................................................................... 72

5.4 MNP Settings .................................................................................................. 73

5.5 Routing Blacklist ............................................................................................. 74

6. SMS .......................................................................................................................... 75

6.1 General ........................................................................................................... 75

6.2 SMS Sender .................................................................................................... 81

6.3 SMS Inbox....................................................................................................... 81

6.4 SMS Outbox.................................................................................................... 82

6.5 SMS Forwarding ............................................................................................. 83

7. Network.................................................................................................................... 84

7.1 LAN Settings ................................................................................................... 84

7.2 WAN Settings ................................................................................................. 85

7.3 VPN Settings ................................................................................................... 85

7.4 DDNS Settings ................................................................................................ 86

7.5 Toolkit ............................................................................................................. 87

7.6 Firewall Settings ............................................................................................. 88

7.7 Security Rules ................................................................................................. 89

7.8 SIP Capture ..................................................................................................... 90

8. Advances .................................................................................................................. 91

8.1 Asterisk API .................................................................................................... 91

8.2 Asterisk CLI ..................................................................................................... 94

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8.3 Asterisk File Editor.......................................................................................... 94

8.4 Internet .......................................................................................................... 95

8.5 Cloud Management ....................................................................................... 95

8.6 Balance ........................................................................................................... 96

8.7 Phone Number ............................................................................................... 97

9. Logs .......................................................................................................................... 98

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1. Overview

1.1 What is VoxStack Series Wireless Gateways？

The VS-GW1202/1600/2120 V2 Wireless Gateway is an upgrade of the

VS-GW1202/1600/2120 series and is now the industry's leading GSM/WCDMA/LTE

VoIP gateway solution for small and medium-sized businesses based on Asterisk.

Thanks to its friendly GUI and unique modular design, users can easily customize the

gateway and perform secondary development via AMI (Asterisk Management

Interface).

The new VoxStack series of gateways are available in three models, the VS-GW1202

V2, the VS-GW1600 V2 and the VS-GW2120 V2 series of wireless gateways. The

modular design of the VS-GW1202 V2 wireless gateway can be plugged into two

GSM/WCDMA/LTE gateway modules, supporting 4 to 8 GSM/WCDMA/LTE channels,

and the VS-GW1600 V2 wireless gateway can be plugged into 5 gateway modules,

supporting 4 to 20 GSM/WCDMA/LTE channels, VS-GW2120 V2 wireless gateway can

insert 11 gateway modules, supporting 4 to 44 GSM/WCDMA/LTE channels. It also

support G.711A, G.711U, G.729 , G.722, G.723, G.726, iLBC, GSM and other coding

and signaling protocols. The new VoxStack series of gateways are seamlessly

connected to 2G/3G/4G networks. The hot-swap design is applied to SIM cards and

GSM/WCDMA/LTE wireless gateway modules, and users can add or remove them as

needed to expand hardware devices.

The VoxStack series of wireless gateways feature a switchboard design that provides

stack ability during hardware upgrades. It supports SMS information transmission

and reception, sending and receiving by email and group sending. The

VS-GW1202/1600/2120 V2 wireless gateway uses standard SIP protocol which is

compatible with mainstream IMS/NGN platforms, IPPBX and SIPS servers, and

supports most VoIP operating system platforms such as Asterisk, Elastix, 3CX,

FreeSWITCH, Broadsoft and so on.

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1.2 Application

Figure 1-2-1 Application Topology

Figure 1-2-2 VS-GW1600 series products positive

Figure1-2-3 VS-GW1600 series front panel

Figure 1-2-4 VS-GW2120 series products positive

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Figure1-2-5 VS-GW2120 series front panel

Table 1-2-1 Status light description

Status Light Color Status

Green and Flash Module Initiating

Red and Flash No SIM Card

Signal Status LED Always red Worst Signal Quality

Always yellow Medium Signal Quality

Always green Best Signal Quality

Flash (0.5s) Communicating

Call Status LED
Blind Normal

Network Status LED Green and Flash Network Connected

Running Status LED Green and Flash(0.5s) Work Normally

Power Indicator Always Green Power on

1.3 Main Features

 Support SIP，IAX2 Protocol

 Add, Modify & Delete SIP/IAX2 Trunk

 SIP/IAX2 Registration with Domain

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 SIP V2.0 RFC3261 Compliance

 DTMF Mode: RFC2833/Inband/SIP Info

 Multiple SIP/IAX2 Registrations modes

 Abundant Codecs:G.711A, G.711U, G.729, G.722, G.723, G.726, GSM

 IPv4，UDP/TCP，DHCP，TELNET，HTTP/HTTPS，TFTP

 PPTP VPN

 HTTP/SSH（Optical Telnet）

 Ping & Traceroute Command on the Web

 Simple Security Strategy: white list, black list, security rules

 Simple and convenient configuration via Web GUI

 Support maintenance and configuration by SSH

 Support configuration files backup and upload

 Support Chinese and English page

 Firmware Update by HTTP

 Support Web and SSH login password modification

 Restore Factory Settings

 CDR(More than 200,000 Lines CDRs Storage Locally)

 System log

 SIP/IAX2 log

 TCP and SIP capture

 Combine Different SIP/IAX2 Trunk into Group

 CLID Display & Hide (Need operators' support )

 Random call interval

 Call Duration Limitation

 Single Call Duration Limitation

 Real Open API Protocol (based on Asterisk)

 Support DISA

 SMSC/SMS/USSD

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 PIN Identification

 Optional Voice Codec

 Ports Group Management

 SMS Remotely Controlling Gateway

 SMS Bulk Transceiver, Sent to Email and Automatically Resend

 SMS Coding/Detecting Automatically Identification

 SMS Forwarding and Quick Reply

 USSD transceiver

 Outbound

 Automatically Reboot

 Support MMP

 Support for custom scripts, dial plans

 Support OpenVox cloud manage

1.4 Physical Information

 Weight:

VS-GW1202 V2: 1300g

VS-GW1600 V2: 3682g

VS-GW2120 V2: 6144g

VS-GWM420G: 162g

VS-GWM420W: 170g

VS-GWM420L: 178g

 Size:

VS-GW1202 V2: 15cm*19cm*4.5cm

VS-GW1600 V2: 44cm*30cm*4.5cm

VS-GW2120 V2: 48.3cm*33.1cm*8.8cm

 Power Consumption:

VS-GW1202 V2: 18W

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VS-GW1600 V2: 46W

VS-GW2120 V2: 88W

 Operation Temperature:0~40°C

 Storage Temperature: -20~70°C

 Operation humidity:10% ~ 90% non-condensing

 WAN: 2*10/100M

1.5 Login

 Default IP: 172.16.98.1

 Username: admin

 Password: admin

For the first time, you can access by using default IP 172.16.98.1. Then configure the

module as you want.

Figure 1-5-1 Login Interface

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2. System

2.1 Status

On the “Status” page, you will find the information of all Modules, SIP, IAX2, Routing

and the Network.

Figure 2-1-1 System Status

Table 2-1-1 Description of System Status

Option Definition

Number of each ports. GSM port starts with "gsm-"，Such as：

Port
gsm-1.1;4G port starts with "lte-"，Such as：lte-2.5.

Signal Display the signal strength of each channels of gateway.

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Band GSM/ LTE.

BER Bit Error Rate.

Carrier Display the network carrier of current SIM card.

Registration
Indicates the registration status of current module.
Status

Post Dial Delay (PDD) is the time from the point at which the

final dialed digit is sent until they hear a ringtone or other

PDD in-band message. In the case where the originating network

needs to play a notification before the call is completed, the

definition of the PDD excludes the duration of such notification.

The Average Call Duration (ACD) is calculated by taking the sum

ACD of billable seconds (bill sec) of answered calls and dividing it by

the number of these answered calls.

Answer Seizure Ratio is a measure of network quality. It’s

calculated by taking the number of successfully answered calls

ASR and dividing by the total number of attempted calls. Since busy

signals and other rejections by the called number count as call

failures, the ASR value can depending on user behavior.

Display the status of the port. “A space” means the port is

Module
unavailable. "INIT" means registering and "READY" means the
Status
port is available.

Remain
Multiply this value by the step size as the limit call time.
Time

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2.2 Time

Table 2-2-1 Definition of Time Settings

Options Definition

System Time Your gateway system time.

The world time zone: Please select the one which is the same
Time Zone
or the closest as your city.

POSIX TZ
Posix time zone strings.
String

Time server domain or hostname. For example:

NTP Server 1
[time.asia.apple.com].

The first reserved NTP server. For example:

NTP Server 2
[time.windows.com].

NTP Server 3 The second reserved NTP server. For example: [time.nist.gov].

Automatic Whether to enable automatic sync time from NTP. On

Sync from NTP (enable), OFF (disable).

Sync from NTP Sync time from NTP server.

Sync from
Sync time from local machine.
Client

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For example, you can configure like this:

Figure 2-2-1 Time Setting

You can synchronize the gateway time in different ways: Sync from NTP or Sync from

Client by pressing different buttons.

2.3 Login Settings

The new gateway has no administration privileges, all you can do is reset a username

and password to manage your gateway. The username and password have all the

permissions to operate the gateway. You can modify “Web Login Settings” and “SSH

Login Settings”. If you have changed these settings, you don’t need to log out, just

rewriting your new username and password will be OK. Also, you can specify the web

server port number. Normally, the default web login mode is "http and https." For

security, you can switch to “only https”.

Table 2-3-1 Definition of Login Settings

Options Definition

Define your username to manage your gateway. Allowed

User Name
characters "-_+. < >&0-9a-zA-Z". Length: 1-32 characters.

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Define your password to manage your gateway Allowed characters

Password
"-_+. < >&0-9a-zA-Z". Length: 1-32 characters.

Confirm
Please input the same password as 'Password' above.
Password

http and https: You can access gateway via link: http://gatewayIP
Login
or https://gatewayIP
Mode
https: You can only access gateway via link: https://gatewayIP

Port Specify the web server port number.

For example, you can configure like this:

Figure 2-3-1 Login Settings

2.4 General

You can choose different languages for your system. If you want to change the

language, you can switch “Advanced” on, then “Download” your current language

package. After that, you can modify the package with the language you need. Then

upload your modified packages, “Choose File” and “Add”. For example:

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Figure 2-4-1 Language Settings

If you switch it on, you can manage your gateway to reboot automatically as you like.

There are four reboot types for you to choose, “By Day, By Week, By Month or By

Running Time”.

Figure 2-4-2 Reboot Type

You can set this enable if you use your system frequently, it can help your system

work more efficiently.

2.5 Tools and Information

Reboot Tools
You can choose system reboot or asterisk reboot separately. The new version of the

gateway retains two file systems. Clicking "System Switch" will reboot the system and

switch to another file system.

Figure 2-5-1 Reboot Tools

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If you press "OK", the system will restart and the current call will be hung up. The

same as the Asterisk Reboot.

Update Firmware
We offer 2 kinds of update types for you, you can choose System Update or System

Online Update.
Figure 2-5-2 Update Firmware

Upload and Backup Configuration

If you want to update your system and remain your previous configuration, you can

backup configuration first. Then you can upload configuration directly. It will be very

convenient for you.

Figure 2-5-3 Upload and Backup Configuration

Voice Record
Select a module to record the voice, the maximum duration of the recording is 3

minutes. When the recording time exceeds 3 minutes, the recording file will be

automatically downloaded.

Figure 2-5-4 Voice Record

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Restore Configuration
Sometimes there is something wrong with your gateway that you don’t know how to

solve it. Mostly you can select factory reset, you just need to press a button and your

gateway will be reset to the factory status.

Figure 2-5-5 Restore Configuration

System Information
On the “Information” page, there shows some basic information about the gateway.

You can see software and hardware version, storage usage, memory usage and some

other useful information.

Figure 2-5-6 System Information

2.6 Setting Wizard

According to the boot, you can change the password, set the time, network, and

create SIP terminals and routes quickly.

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Figure 2-6-1 Setting Wizard

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3. Module
You can see the information of SIM card on this page.

3.1 Module Settings

Figure 3-1-1 Module Settings

On this page, you can see the information of the module status, click the
button to configure the port.

Figure 3-1-2 Port Configuration

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You can choose different bands in the options.

Figure 3-1-3 Band binding

If you have set your Pin Code, you can check on like this:
Figure 3-1-4 PIN Code Application

Then enter the password and the system will recognize the number of the SIM card.

It can help you prevent the SIM card from being stolen and increase the safety factor.

If you want to hide your number when you call out, you can just switch CLIR “ON” (Of

course you need your operator’s support).

Figure 3-1-5 CLIR Application

OpenVox GSM/3G Gateway supports optional GSM voice codec. For more details,

you can see the picture below：

Figure 3-1-6 GSM Voice Codec

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We also provide IMEI automatically modified performance.

Figure 3-1-7 Automatic IMEI modification

If you want to modify the IMEI number, log in to the gateway to modify the IP

address as follows. Input http://gatewayIP/cgi-bin/php/gsm-autoimei.php in your

browser. Then log in to the web page and you will see the setting as “Enable” as “ON”

as shown below. Otherwise, the IMEI number cannot be modified.

Figure 3-1-8 IMEI Modification

You can choose to modify one, more ports or all ports, and you can set the automatic

modification interval by filling in the required time.

Figure 3-1-9 Time Interval

If you select "Modify IMEI now", the changes will take effect immediately. If you

select "Force", the system will hang up the current call and modify the IMEI.

Click the button to set it. There are two ways to modify

the IMEI: manual modification or automatic generation.

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Figure 3-1-10 Advanced Setting

As you can see, each port is set to any number. "X" means any number from 0-9. All

you have to do is fill in all the lines and click "Set All". Click "Save", the current IMEI

will change, this is automatically generated.

If you want to set the IMEI to a specific value, click on “Modify” and enter a new IMEI

as required.

Figure 3-1-11 Manually set IMEI

After the configuration is completed, click "Back Home" to return to the gateway

interface.

Table 3-1-1 Definition of Module Settings

Options Definition

The alias of the each port. Input name without space here.

Name Allowed characters ： "-_+.<>&0-9a-zA-Z".Length: 1-32

characters.

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Speaker The range of the speaker volume level is 0-100. This will

Volume adjust the loud speaker volume level by an AT command.

Microphone The range of the microphone volume is: 0-15. This will

Volume change the microphone volume level by an AT command.

The number prefix of the outbound call through the GSM

Dial Prefix
channel.

Personal identification numbers of SIM card. PIN code can be

PIN Code
modified to prevent SIM card from being stolen.

Custom AT
Use “|” to split AT command when user use custom AT
commands
commands to start system.
when start

Caller ID restriction, this function is used to hide the caller ID

of the SIM card number. The gateway will add ‘#31#’ in front
CLIR
of the mobile number. This function must be operated by a

professional operator.

SMS Center
Your SMS center number of your local carrier.
Number

Module IMEI Press the Modify button to automatically modify

Save settings to other ports:

Figure 3-1-12 Save to port

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Your call status will be displayed on the main screen.

Figure 3-1-13 Module Information

3.2 Call Forwarding

Sometimes it's not convenient for you to answer the call. If you don't want to lose

some important calls, you can choose call forwarding. There are different kinds of call

types for you to choose, Such as call forwarding unconditional, call forwarding no

reply, call forwarding busy and call forward on not reachable. If you want to cancel

the call forwarding settings, you can choose to cancel all.

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Figure 3-2-1 Call Forwarding

3.3 Call Waiting

You can switch the Call Waiting Function of the SIM card on/off in the port on this

page.

Figure 3-3-1 Call Waiting

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Table 3-3-1 Definition of Call Waiting

Options Definition

Switch on call Choose the port you want to set, switch to "On" and click the

waiting "Settings" button.

Switch off call Choose the port you want to set, switch to "Off" and click the

waiting "Settings" button.

Status It will display the results of your execution.

Settings Used to switch the call waiting function on or off.

Used to query the status of call waiting, observe whether the

Query
port that sets the call waiting is open or closed.

3.4 DTMF

You can do some DTMF Detection Settings if you choose “MODULE –> DTMF”.

Figure 3-4-1 DTMF Detection Settings

Notice: You don’t have to modify these settings if you don’t have special need. You

can just choose “Default”.

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Table 3-4-1 Definition of DTMF Detection Settings

Options Definition

It is the power difference between the row and the column

DTMF Normal
energies. Normal Twist is where the Column energy is greater
Twist and
than the Row energy. Reverse Twist is where the Row energy is
Reverse Twist
greater.

The smaller the value, the easier the detection. If you lost
DTMF Relative
some numbers, you can try to put the value down. The
Peak Row
adjustment range is 0.02 at a time.

The smaller the value, the easier the detection. If you lost
DTMF Relative
some numbers, you can try to put the value down. The
Peak Col
adjustment range is 0.1 at a time.

DTMF Hits
Simple match value, choose 2 or 3.
Begin

The time interval between the two numbers you entered.

DTMF Misses
Adjust the speed of input. The smaller value represents the
End
shorter intervals.

3.5 BCCH

Figure 3-5-1 BCCH Settings

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The BCCH mode can be changed by clicking the button.

Figure 3-5-2 BCCH Mode

3.6 Toolkit

You can get USSD information, send AT command or check the number of this

module. The AT command is very helpful for you when debugging a module.

Figure 3-6-1 Function Options

Table 3-6-1 Description of Definition of Functions

Options Definition

Enter a specific USSD number (For example,*142# to check

your SIM card's balance. This USSD number is might be

Get USSD
different from different carriers) to get the USSD information.

The gateway will try to get by AT commands.

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To perform some specific AT commands. It is useful when you

have a debug of the modem. For example, perform [AT+CSQ]

AT Command
to check what signal qualify it is. In AT commands, there is no

difference between “a” and “A” (case insensitive).

Enter a known number (like your mobile phone number) to

check what number it is of the SIM card. Click "Execute", then

Check Number the gateway will dial to the number you already input. It only

rings for one time and hangs up at once. Not generating

telephone charge during this procedure.

If you want to send an AT command, enter the command, select a port first and then

select "Copy to selected ", and finally click "Execute".

Figure 3-6-2 AT Command Example

3.7 Module Update

You can update the module version or update the microcontroller by yourself on this

page.

Figure 3-7-1 Module Update

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3.8 Call and SMS Limit

Click the edit button to set call restrictions, lock cards, and SMS restrictions. You can

also view statistics on call/sms sent here.

Figure 3-8-1 Call and SMS Limit

Call Limit
You can limit the number of daily calls, number of days connected, and number of

hours of calls for the selected channel.

Figure 3-8-2 Call Limit

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Call Limit Time

Now we offer you two types of call duration limit, you can choose “Single Call

Duration Limit” or “Call Duration Limitation” to control your calling time.

This will limit the time of each call. First you need to switch “Enable” on, then you

can set “Step” and “Single Call Duration Limitation” any digits you want. When you

make a call through this port, your calling time will be limited.

Single Call Duration Limitation：If your calling time exceeds the set value, the system

will hang up the call. Multiplying the step size by a single call time is to allow a single

call duration.

Figure 3-8-3 Single Settings

Call Duration Limitation：This will limit your total calling time of this port. If remain

time is 0, it will not send calls through this port.

Figure 3-8-4 Call Duration Limitation Settings

The same as the algorithm for the single call duration limit, the total call time of the

port cannot exceed the product of the "Step" and the "Call Duration Limitation". The

Minimum Billing Time value must be less than the step size.

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You can set a value for the Alarm Threshold. When the remaining call duration of the

port reaches the set value, the gateway will send alarm info to the designated phone.

You can also turn on enable auto reset, which can be selected from one day, one

week or one month. After the automatic reset time is reached, the remaining call

duration is reset to the set total call duration.

Figure 3-8-5 Auto Reset Settings

Table 3-8-1 Description of Call Duration Limit Settings

Options Definition

The value range of the step length is 1-999s. Multiplying the step
Step
size by a single call time is to allow a single call duration.

Enable Single The value of the single call duration limit, ranging from 1-999999.

Call Duration The step size multiplied by the single talk time is the single call

Limit duration.

Enable Call The value of the total call duration limit, in the range of 1-999999.

Duration The step size multiplied by the total talk time is the total call

Limitation duration.

Minimum
After a period of conversation, the operator starts billing, in
Charging
seconds. This value must be less than the step size.
Time

The threshold for the duration of the call. When the duration of
Alarm
the call is equal or less than this value, the gateway will send an
Threshold
alert message to the specified phone number via SMS (only once

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before the total call duration is reset). This value must be less

than the call duration limit.

Alarm Phone User will received alarm message from the gateway when

Number receiving alarm phone number.

Alarm The alarm port information description sends the alarm

Description information to the user's phone.

Remaining Multiplying this value by the step size is the remaining call

call duration duration.

Enable Auto The remaining talk time is automatically restored, that is, the total

Reset number of minutes of the call is restored.

Auto Reset
Reset call minutes by date, by week, by month.
Type

Next Reset Define the next reset date. The system will count from that date

Time and be set as a reset period.

Lock Card
The lock card detection switch is a switch function of the lock card. After opening,

the call failure lock card condition parameter needs to be set. After the lock card

condition is reached, the SIM card is disabled and cannot be allocated for use unless

the card is removed, the gateway is restarted, and the manual is unlocked (Duration

limit requires manual reset), turn off the lock card function, etc.

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Figure 3-8-6 Lock Card

Table 3-8-2 Description of Lock Card

Options Definition

Call Failed Mark The number of consecutive calls failed to reach the number

Count of settings, making the port.

Lock the port after the number of successive call failures

Called Failed Lock
reaches the set number, restrict the port expiration, and no
Count
longer select the port expiration in the group policy.

After opening, when the number of successive call failures

reaches the set value, send message to check whether the

SMS Send
port is available; if the message send successfully, it will clear
Detection Switch
the number of successive call failures; if the message fails, it

will limit the port’s outgoing.

When closed, the successful sending of message indicates

Testing SMS that the port is available; when opened, the successful

report sending of message and the receipt of message report

indicate that the port is available.

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SMS Limit
Figure 3-8-7 SMS Limit

Table 3-8-3 Definition of SMS Limit

Options Definition

When closed, no matter whether the message is sent

SMS Limit successfully or not, the number of messages is counted；When

Success Flag opened, the number of messages is counted when the message

is sent successfully.

Day Limit SMS Limit the number of SMS sent daily, the default value is 0,

Count indicating unlimited.

Month Limit Limit the number of SMS sent per month, the default value is 0,

SMS Count indicating unlimited.

SMS Clean The number of messages sent monthly is automatically cleared

Date at 0 points, 0 minutes and 0 seconds on the set date.

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4. VOIP

4.1 VOIP Endpoints

This page shows everything about your SIP&IAX2, you can see the status of each

SIP&IAX2.
Figure 4-1 SIP&IAX2 Endpoints

4.1.1 Add New SIP Endpoint

Main SIP Endpoint Settings:

You can click button to add a new SIP endpoint, and if you

want to modify existed endpoints, you can click button.

There are 3 kinds of registration types for you to choose: None, Server or Client. You

can configure as follows:

If you set up a SIP endpoint by registration “None” to a server, then you can’t register

other SIP endpoints to this server. (If you add other SIP endpoints, this will cause

Out-band Routes and Trunks confused.)

Figure 4-1-1 None Registration

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For convenience, we have designed a method that you can register your SIP endpoint

to your gateway, thus your gateway just work as a server.

Figure 4-1-2 Server Registration

Also you can choose registration by “This gateway registers with the endpoint”, it’s

the same with “None”, except the name and the password.

Figure 4-1-3 Client Registration

Table 4-1-1 Definition of SIP Options

Options Definition

A name is easier to understand, only for reference, does not

Name
participate in SIP authentication.

Username Register name in your SIP server.

Password Authenticating with the gateway and characters are allowed.

Whether the terminal registers with the gateway or the gateway

Registration
registers with the terminal.

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None --- Not registering;

Server --- When register as this type, it means the gateway acts

as a SIP server, and SIP endpoints register to the gateway;

Client --- When register as this type, it means the gateway acts as

a client, and the endpoint should be register to a SIP server.

IP address or hostname of the endpoint or 'dynamic' if the

Hostname or
endpoint has a dynamic IP address. This will require registration.
IP
Notice: If you entered the domain name and changed the DNS,
Address
you must restart asterisk.

This sets the possible transport types for outgoing. Order of

usage, when the respective transport protocols are enabled, is

UDP, TCP, TLS. The first enabled transport type is only used for
Transport
outbound messages until a Registration takes place. During the

peer registration the transport type may change to another

supported type if the peer requests so.

No --- Use report if the remote needs it.

Force Report on—Force report on.

Yes --- Force the report to open and perform commedia RTP
NAT Traversal
processing.

Report if requested and commedia --- If the remote needs it, use

report and perform commedia RTP processing.

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Advanced: Registration Options

Figure 4-1-4 Advanced Registration Options

Table 4-1-2 Definition of Registration Options

Options Definition

Authentication
Just a username used when registering。
User

When Gateway registers as a SIP user agent to a SIP

Register
proxy (provider), calls from this provider connect to this
Extension
local extension.

Register user name, it is the user of register => u

Register User
ser[:secret[:authuser]]@host[:port][/extension].

e.g. When the Contact User is 402

Contact User
Contact: <sip:[email protected]:5060;transport=UDP>

From User A username to identify the gateway to this endpoint.

From Domain A domain to identify the gateway to this endpoint.

Port The port number used by the gateway to link the peer

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Qualify Whether to check the link status of the peer

How often, in seconds, to check the endpoint's

Qualify Frequency
connection status.

A proxy to which the gateway will send all outbound

Outbound Proxy signaling instead of sending signaling directly to

endpoints.

After opening, customers can customize their own

Custom Registry
registry.

Customers can enter their own registration informa

tion. The format is:：user[:secret[:authuser]]@host[:

Registry String
port][/extension]

e.g.：[email protected]:[email protected]/1001

Enable Outbound
Outbound proxy to Host On / Off.
proxy to Host

Call Settings
Figure 4-1-5 Call Settings

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Table 4-1-3 Description of DTMF Detection Settings

Options Definition

Set default DTMF Mode for sending DTMF. Default: rfc2833.

DTMF Mode Other options: 'info', SIP INFO message(application/dtmf-relay);

'Inband', Inband audio(require 64kbit codec -alaw, ulaw).

Trust
Whether or not the Remote-Party-ID header should be trusted.
Remote-Party-ID

Send
Whether or not to send the Remote-Party-ID header.
Remote-Party-ID

Remote-Party-ID How to set the Remote-Party-ID header: from Remote-Party-ID

Format or from P-Asserted-Identity.

Caller ID
Whether or not to display the Caller ID.
Presentation

Usually used when this sip works as a trunk. Limit the maximum
Call Limit
number of channels supported by sip trunks.

Advanced: Signaling Settings

Figure 4-1-6 Signaling Settings

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Table 4-1-4 Definition of Signaling Options

Options Definition

Whether the tone is ringing.

Never：Indicates that the incoming calls are never applied.

Progress Inband
Option values: yes, no, never.

Default: never.

Append
Whether or not to add‘user=phone’to URIs that contain a
user=phone to
valid phone number.
URI

Add Q.850 Whether or not to add a reason header and use it if is

Reason Headers available.

By default, the gateway will honor the session version

number in SDP packets and will only modify the SDP session

if the version number changes.

Ignore SDP Switch this option off to force the gateway to ignore the SDP

Version session version number and treat all SDP data as new data.

This is required for devices that send non-standard SDP

packets (observed with Microsoft OCS).

By default this option is on.

The value of parameter direct media is one of them (no, yes,

Direct media nonat, update, outgoing).

Default value is yes.

Whether or not to globally enable transfers.

Allow Transfers Choosing 'no' will disable all transfers (unless enabled in

peers or users).

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Default is enabled.

Whether or not to allow 302 or REDIR to non-local SIP

Allow
address.
Promiscuous
Notice：Redirecting the local system causes a loop call, which
Redirects
is not supported by asterisk.

Max Forwards Setting for the SIP Max-Forwards header (loop prevention).

Send TRYING on
Send a 100 Trying when the endpoint registers.
REGISTER

Advanced: Timer Settings

Figure 4-1-7 Timer Settings

Table 4-1-5 Definition of Timer Options

Options Definition

This timer is used primarily in INVITE transactions. The default

Default T1 for Timer T1 is 500 ms or the measured run-trip time between

Timer the gateway and the device if you have qualify=yes for the

device.

Call Setup If a provisional response is not received in this amount of time,

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Timer the call will auto congest.

The default is 64*T1.

There are three modes for you to choose: originate, request and

run session-timers always; Accept or run the session timer only

Session Timers
when requested by another user agent；Refuse, Do not run

session timers in any case.

Minimum

Session Minimum session refresh interval in seconds. The default is 90

Refresh secs.

Interval

Maximum

Session Maximum session refresh interval in seconds. The default is

Refresh 1800secs.

Interval

Session
The session refresher, UAC or UAS. The default is UAS.
Refresher

4.1.2 Add New IAX2 Endpoint

You can click button to add a new IAX2 endpoint and if you

want to modify existed endpoints, you can click button.

There are three registration types: None, Server or Client.

You can configure as follows:

If you set up an IAx2 endpoint by registration “None” to a server, then you can’t

register other IAX2 endpoints to this server, just authenticate the username and

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password.

Figure 4-1-8 None Registration

For convenience, you can choose registration by “Server”, thus your gateway just

work as a server.
Figure 4-1-9 Server Registration

Also you can choose registration by “Client”, it will work as a Client.

Figure 4-1-10 Client Registration

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Table 4-1-6 Definition of IAX2 Options

Options Definition

A name which is able to read by human. And it's only used for
Name
user's reference.

The endpoint will use username to authenticate with the

Username
gateway.

The endpoint will use password to authenticate with the

Password
gateway.

Whether this endpoint will register to this gateway or this

gateway to the endpoint.

None --- Not registering;

Endpoint registers with this gateway --- When register as this

Registration type, it means the gateway acts as a IAX2 server, and IAX2

endpoints register to the gateway;

This gateway registers with the endpoint --- When register as

this type, it means the gateway acts as a IAX2 client, and the

endpoint should be register to a IAX2 server;

IP address or hostname of the endpoint or 'dynamic' if the

Hostname
endpoint has a dynamic IP address. This will require registration.
or
Notice: If you entered the domain name and changed the DNS,
IP Address
you must restart asterisk.

There are three authentication methods that are supported:

md5, plaintext and rsa. The least secure is "plaintext", which

Auth
sends passwords clear text across the net. Uses a

challenge/response md5 sum arrangement, but still requires

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both ends have plain text access to the secret. "rsa" allows

unidirectional secret knowledge through public/private keys.

Transfer Disable or not IAX2 native transfer .

Trunk Use IAX2 trunking with this host.

Advanced: Registration Options

Figure 4-1-11 Registration Options

Table 4-1-7 Definition of Registration Options

Options Definition

Qualify Whether or not to check the endpoint's connection status.

Qualify Use the average of the last two connection results to reduce

Smoothing the lag end of the misdetection. The default is no.

How frequently to ping the peer when everything seems to be

Qualify Freq Ok
OK, in milliseconds.

Qualify Freq How frequently to ping the peer when it is either, LAGGED or

Not Ok UNAVAILABLE, in milliseconds.

The port number the gateway will connect to at this endpoint.

port
Default is 4569.

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IAX2 Encryption
Figure 4-1-12 IAX2 Encryptions

Table 4-1-8 Definition of Encryption Options

Options Definition

Encryption Enable IAX2 encryption. The default is no.

Force encryption insures no connection is established unless

both sides support encryption. By switching this option on,

Force Encryption
encryption is automatically switch on as well. The default is

no.

IAX2 Trunk Settings

Figure 4-1-13 IAX2 Trunk Settings

Table 4-1-9 Definition of Trunk Options

Options Definition

The default is 128000 bytes, which supports up to 800 calls

Trunk Max Size
of ulaw at 20ms a frame.

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Trunk MTU Set the maximum transmission unit for IAX2 UDP trunking.

How frequently to send trunk messages in msec. This is

Trunk Frequency
20ms by default.

Ensure that frame timestamps get sent end-to-end

Trunk Time Stamps
properly.

Minimum amounts of time that IAX2 peers can request as a

Min. RegExpire
registration expiration interval in seconds.

Maximum amounts of time that IAX2 peers can request as a

Max. RegExpire
registration expiration interval in seconds.

4.2 Batch SIP Endpoints

On this page, you can create multiple SIP Endpoints at the same time.

Figure 4-2-1 Multiple SIP Endpoints Settings

You should select the number of SIPS to be created after you filling in the username,

password, domain name or IP address, port and registration mode in the first line.

You can create up to the same number of SIP endpoints as the number of device

ports at a time；Then choose whether to tick “AutoPassword” button。After the

above configuration is completed, click Batch Settings, and then save to create SIP

endpoints in batches.

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Table 4-2-1 Definition of Multiple SIP Endpoints

Options Definition

The display name is the same as the registered name on the SIP
Username
server by default.

Password Authenticating with the gateway and characters are allowed.

Hostname or
IP address or hostname of the endpoint or 'dynamic' if the
IP
endpoint has a dynamic IP address. This will require registration.
Address

Port Port number registration.

None --- Not registering;

Server --- When register as this type, it means the gateway acts

Registration as a SIP server, and SIP endpoints register to the gateway;

Client --- When register as this type, it means the gateway acts

as a client, and the endpoint should be register to a SIP server.

Tick - Automatically increments based on the password entered

in the first line.

AutoPassword
Do not tick- All SIP endpoints have the same password as the first

one.

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4.3 Advanced SIP Settings

Networking
Figure 4-3-1 Networking General

Table 4-3-1 Definition of Networking General Options

Options Definition

UDP Bind Port Choose a port on which to listen for UDP traffic.

Enable TCP Enable server for incoming TCP connection (default is no).

TCP Bind Port Choose a port on which to listen for TCP traffic.

TCP The maximum number of seconds a client has to authenticate.

Authentication If the client does not authenticate before this timeout expires,

Timeout the client will be disconnected (default value is: 30 seconds) .

TCP
The maximum number of unauthenticated sessions that will be
Authentication
allowed to connect at any given time (default is:50).
Limit

Enable DNS SRV lookups on outbound calls.

Enable
Notice: The gateway uses the first host in the SRV record. This
Hostname
function can be used in dial-up activation to dial SIP calls on
Lookup
the Internet via domain name.

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Whether enable the internal SIP calls or not when you select
Enable Internal
the registration option "Endpoint registers with this gateway".
SIP Call

Internal SIP Call

Specify a prefix before routing the internal calls.
Prefix S

Figure 4-3-2 NAT Settings

Table 4-3-2 Definition of NAT Settings Options

Options Definition

Format:192.168.0.0/255.255.0.0 or 172.16.0.0./12.

A list of IP address or IP ranges which are located inside a NAT

Local Network network. This gateway will replace the internal IP address in SIP

and SDP messages with the external IP address when a NAT

exists between the gateway and other endpoints.

Local Network
Add local IP address list.
List

Subscribe Through the use of the test_stun_monitor module, the gateway

Network has the ability to detect when the perceived external network

Change Event address has changed. When the stun_monitor is installed and

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configured, chan_sip will renew all outbound registrations

when the monitor detects any sort of network change has

occurred. By default this option is enabled, but

res_stun_monitor only takes effect once. If res_stun_monitor is

enabled and you don't want changes on the network to cause

an outbound registration, use the option below to disable this

feature.

Match External Only substitute the extern address or extern host setting if it

Address Locally matches.

Dynamic Dynamic hosts are not allowed to register with the static host's

Exclude Static IP address, which will avoid the same IP registration error.

Externally
The externally mapped TCP port, when the gateway is behind a
Mapped TCP
static NAT or PAT.
Port

The external address (and optional TCP port) of the NAT.

External Address = hostname[:port].

External Specifies a static address[:port] to be used in SIP and SDP

Address messages. Examples:

External Address = 12.34.56.78

External Address = 12.34.56.78:9900.

How often to perform a hostname lookup. This can be useful

Hostname when your NAT device lets you choose the port mapping, but

Refresh Interval the IP address is dynamic. Beware, you might suffer from

service disruption when the name server resolution fails.

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Figure 4-3-3 RTP Settings

Table 4-3-3 Definition of RTP Settings Options

Options Definition

Start of RTP Port Range Start of port numbers to be used for RTP.

End of RTP port Range End of port numbers to be used for RTP.

Parsing and compatibility

Figure 4-3-4 Parsing and Compatibility

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Table 4-3-4 Instruction of Parsing and Compatibility

Options Definition

Strict RFC Check header tags，character conversion in URIs, and multiline

Interpretation headers for strict SIP compatibility(default is yes).

Send Compact
Send compact SIP headers.
Headers

Allows you to change the username filed in the SDP owner

SDP Owner
string. This filed MUST NOT contain spaces.

When a dialog is started with another SIP endpoint, the other

endpoint should include an Allow header telling us what SIP

methods the endpoint implements. However, some endpoints

Disallowed SIP either do not include an Allow header or lie about what

Methods methods they implement. In the former case, the gateway

makes the assumption that the endpoint supports all known

SIP methods. If you know that your SIP endpoint does not

provide support for a specific method, then you may provide a

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list of methods that your endpoint does not implement in the

disallowed methods option. Note that if your endpoint is

truthful with its Allow header, then there is no need to set this

option.

The Shrink Caller Id function removes '(', ' ', ')', non-trailing '.',

and '-' not in square brackets. For example, the caller id value
Shrink Caller ID
555.5555 becomes 5555555 when this option is enabled. By

default this option is on.

Default：SIP From and Number.

For example：When selecting SIP From, Name is Peter an

Caller ID d Number is 402.

The From mode is: "Peter"<sip:[email protected];transport

=UDP>;tag=bd481672.

Default: EXTEN

For example: When selecting SIP To, Name is Jason and

Callee ID Number is 401.

The mode is: "Jason"<sip:[email protected];transport=UD

P>.

Maximum
Maximum allowed time of incoming registrations and
Registration
subscriptions (seconds).
Expiry

Minimum

Registration Minimum length of registrations/subscriptions (default 60).

Expiry

Default Default length of incoming/outgoing registration.

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Registration

Expiry

Registration How often, in seconds, to retry registration calls. Default 20

Timeout seconds.

Number of
Number of registration attempts before we give up.
Registration

Security
Figure 4-3-5 Security Settings

Table 4-3-5 Instruction of Security

Options Definition

Match Auth If available, match user entry using the 'username' field from the

Username authentication line instead of the 'from' field.

Realm for digest authentication. All realms must be globally

Realm unique according to the RFC3261 standard and can generally be

set to hostname or domain name.

Use Domain
Use the domain from the SIP Domains setting as the realm.
as Realm

When an INVITE or REGISTER request is rejected for any reason,

Always Auth
the same reason is always used, the username is legal but the
Reject
password is incorrect. It does not tell the requester whether

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there is this user or peer, which will reduce the possibility of the

attacker scanning the SIP account. This parameter is enabled by

default.

Authenticate
Enabling this option will authenticate OPTIONS requests just like
Options
INVITE requests are. By default this option is disabled.
Requests

Allow or reject customer calls (default yes, allowed). If your

Allow Guest gateway is connected to the Internet and allows customers to

Calling call, you want to check which services are available to everyone,

by enabling them in the default context.

Media
Figure 4-3-6 Media Settings

Table 4-3-6 Instruction of Media

Options Definition

TOS for SIP Packets Sets type of service for SIP packets.

TOS for RTP Packets Sets type of service for RTP packets.

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Codec Settings
Select codecs from the list below.
Figure 4-3-7 Codec Settings

4.4 Advanced IAX2 Settings

General Settings
Figure 4-4-1 General Settings

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Table 4-4-1 Instruction of General

Options Definition

Bind Port Bind port and bindaddr may be specified

Set Iaxcompat to yes if you need to use a tiered switch, or if

Enable some of the latency may occur when performing a lookup in a

IAXCompat dial plan, otherwise enabling it will result in a small performance

conflict.

Disable UDP checksums (if Nochecksums is set, then

Enable
Nochecksums will be calculated/checked on systems supporting
Nochecksums
this feature).

ADSI (Analog Display Services Interface) can be enabled if you

ADIS
have (or may have) ADSI compatible CPE equipment.

SRV Lookup Whether or not to perform an SRV lookup on outbound calls.

You may specify a global default AMA flag for IAX internal calls.
AMA Flags
These flags are used in the generation of call detail records.

If we don't get ACK to our NEW within 2000ms, and Auto Kill is

set to yes, then we cancel the whole thing (that's enough time

Auto Kill for one retransmission only). This is used to keep things from

stalling for a long time for a host that is not available, However,

this function is not friendly in the case of poor connection status.

You may specify a global default language for users. This can be

Language specified also on a per-user basis. If you ignore this item, it will

use the default English language.

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You may specify a default account for Call Detail Records (CDRs)
Account Code
in addition to specifying on a per-user basis.

Music on Hold
Figure 4-4-2 Music on Hold Settings

Table 4-4-2 Instruction of Music on Hold

Options Definition

Specifies the type of waiting music that plays while the

Mohsuggest channel is on hold. It can specify global settings or be

specified by the user or each peer.

Set the content of the channel playback when the phone is

suspended. If it is "default", the music specified in the

Mohinterpret dialing rules will be played in the channel when the phone

hangs. If it is "transfer", it will be replaced by a message

signal.

Instruction of Codec Settings

Figure 4-4-3 Codec Settings

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Table 4-4-3 Instruction of Codec Settings

Options Definition

Specify bandwidth of low, medium, or high to control which

Band Width
codes are used in general.

Disallow Adjust the type of codec to be switched off.

Allow Adjust the type of codec to be switched on and the priority.

Codec priority controls the codec negotiation of an inbound

IAX2 call. This option is inherited to all user entities. It can

Codec Priority
also be defined in each user entity separately which will

override the setting in general.

Jitter Buffer Settings

Figure 4-4-4 Jitter Buffer

Table 4-4-4 Instruction of Jitter Buffer

Options Definition

Whether to set a global buffer should be enabled by an

unstable network environment, generally you do not need.

Jitter Buffer
Because the terminal device buffers the jitter processing.

The default is "no".

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In the ideal world, when we bridge VoIP channels we don’t

want to jitter buffering on the switch, since the endpoints

Force Jitter Buffer can each handle this. However, some endpoints may have

poor jitter buffers themselves, so this option will force to

always jitter buffer, even in this case. The default is “no”.

Max Jitter Buffers A maximum size for the jitter buffer.

When the jitter buffer notices a significant change in delay

that continues over a few frames, it will resync, assuming

that the change in delay was caused by a timestamping

Resyncthreshold mix-up. The threshold for noticing a change in delay is

measured as twice the measured jitter plus this resync

threshold. Resyncing can be disabled by setting this

parameter to -1.

The maximum number of interpolation frames the jitter

Max Jitter Interps
buffer should return in a row

Number of milliseconds by which the new jitter buffer will

pad its size. The default is 40, so without modification, the

Jitter Target Extra new jitter buffer will set its size to the jitter value plus 40

milliseconds. Increasing this value may help if your network

normally has low jitter, but occasionally has spikes.

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Misc Settings
Figure 4-4-5 Misc Settings

Table 4-4-5 Instruction of Misc Settings

Options Definition

IAX Thread Count Establishes the number of IAX helper threads to handle I/O

IAX2 Max Thread Establishes the number of extra dynamic threads that may be

Count spawned to handle I/O.

The 'Max Call Numbers' option limits the amount of call

numbers allowed for each individual remote IP address. Once

an IP address reaches its call number limit, no more new

Max Call Number connections are allowed until the previous ones close. This

option can be used in a peer definition as well, but only takes

effect for the IP of a dynamic peer after it completes

registration.

The 'MaxCallNumbers_Nonvalidated' is used to set the

combined number of call numbers that can be allocated for

MaxCallNumbers_ connections where call token validation has been disabled.

Nonvalidated Unlike the ' Max Call Number ' option, this limit is not

separate for each individual IP address. Any connection

resulting in a non-call token validated call number being

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allocated contributes to this limit. For use cases, see the call

token user guide. This option's default value of 8192 should

be sufficient in most cases.

Quality of Service
Figure 4-4-6 Quality of Service

Table 4-4-6 Instruction of Quality of Service

Options Definition

TOS Type of service.

COS Class of service.

4.5 SIP Account Security

The SIP account security module greatly enhances the security of the system. The

wireless gateway supports TLS encrypted calls (sip), which requires sip phone

support.
Figure 4-5-1 SIP Account Security

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You need to understand its rules when using TLS. The following table shows the

setting parameters of TLS.

Table 4-5-1 Parameters of TLS settings

Options Definition

TLS Enable Enable or disable DTLS-SRTP support.

If set it off, there is no need to verify the server certificate

when the client is operating. If you do not have a server CA

TLS Verify Server
certificate, you can disable authentication to connect

without the need for a TLS CA file.

Port TLS port SIP registration, default port 5061.

Values include tlsv1, sslv3, sslv2, which specify the protocol

TLS Client Method
for the outbound client connection. The default is sslv2.

Enter the key name, IP address, organization and password in the above settings to

create a ca certificate. After mutual authentication between the client and the server,

it can be accessed with a certificate.

We need to do a few basic steps:

1. Your asterisk server requires an authentication. We must create or add a

certificate on the asterisk server. We need to create a digital key for our server,

the server key is the key.pem file, and the certificate request is request.pem.

2. Add some configuration settings to the sip.conf file.

3. Configure the client to use TLS.

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5. Routing

5.1 Call Routing Rules

Figure 5-1-1 Call Routing Rules

You are allowed to set up new call routing rules by , and after

setting routing rules, move rules’ order by pulling up and down, click

button to edit the routing and to delete it. Finally click the button
to save what you set.

Call Routing Rules

You can click button to set up your routings

Figure 5-1-2 Call Routing Rules

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The figure above shows that all the phones in the group ALL are transferred to the

IAX2-2133. When the "incoming source" is GSM/LTE port, "prepend", "prefix" and

"match pattern" in the "Advanced Routing Rules" are invalid, and only the "CallerID"

option is available.

Table 5-1-1 Definition of Routing Options

Options Definition

The name of this route. Should be used to describe what

Routing Name types of calls this route matches (for example:

‘sip1TOport1’or‘port1TOsip1’).

Call Comes in
The launching point of incoming calls.
From

Send Call Through The destination to receive the incoming calls.

Figure 5-1-3 DISA Settings

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In the DISA Settings, you can set a Password, Authentication or Secondary Dialing.

Figure 5-1-4 Advance Routing Rule Settings

In the Advance Routing Rule Settings, you can set the function of the dial Patterns、

Time Patterns、Calling, Forward, Delay and Failover.

Table 5-1-2 Description of Advanced Routing Rule

Options Definition

A Dial Pattern is a unique set of digits that will select this

route and send the call to the designated trunks. If a dialed

pattern matches this route, no subsequent routes will be

tried. If Time Groups are enabled, subsequent routes will be

checked for matches outside of the designated time(s).

Dial Patterns that Rules:

will X matches any digit from 0-9

use this Route Z matches any digit from 1-9

N matches any digit from 2-9

[1237-9] matches any digit in the brackets (for example:

1,2,3,7,8,9).

Wildcard: matches one or more dialed digits.

Prepend: Digits to prepend to a successful match.

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If the dialed number matches the patterns specified by the

subsequent columns, then this will be prepended before

sending to the trunks.

Prefix: Prefix to remove on a successful match.

The dialed number is compared to this and the subsequent

columns for a match. Upon a match, this prefix is removed

from the dialed number before sending it to the trunks.

Match pattern: The dialed number will be compared against

the prefix + this match pattern. Upon a match, the match

pattern portion of the dialed number will be sent to the

trunks.

CallerID: If CallerID is supplied, the dialed number will only

match the prefix + match pattern if the CallerID has been

transmitted matches this.

When extensions make outbound calls, the CallerID will be

their extension number and NOT their Outbound CID.

The above special matching sequences can be used for

CallerID matching similar to other number matches.

Set the Caller ID What caller ID name would you like to set before sending

Name to this call to the endpoint.

Set the Caller ID What caller ID number would you like to set before sending

Number to this call to the endpoint.

This number will be sent to target PBX or SIP Server as a DID

Forward Number
number for inbound rule settings.

Custom Context Custom the context of dialing rules.

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Random Delay Randomly obtain any value within the threshold interval.

Failover Call The gateway will attempt to send the call out each of these

Through Number in the order you specify.

5.2 Groups

Sometimes you want to make a call through one port, but you don’t know if it is

available, so you have to check which port is free. That would be troublesome. But

with our product, you don’t need to worry about it. You can combine many Port or

SIP to groups. Then if you want to make a call, it will find available port automatically.

For the "Ascending" policy, if 2 or more port members are selected, it will use the

first available port to call/access the call. In this case, if the gsm-1.1 is available, it will

always use the gsm-1.1 call, otherwise, it will use gsm-1.2, and so on.
Figure 5-2-1 Routing Groups

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5.3 Batch Creating rules

This page can generate multiple routing rules at the same time.

Figure 5-3-1 Batch Creating rules

The SIM number will be sent to the target PBX or SIP server as DID number, and the

calling number will be set to the caller ID in the outgoing route.

Figure 5-3-2 Create an incoming route

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Figure 5-3-3 Create an outgoing route

5.4 MNP Settings

Mobile Number Portability allows switching between mobile phone operators

without changing the mobile number.

Figure 5-4-1 MNP Settings

The URL is displayed as a password string. So please enter the URL in the format

below and check if it is correct. After confirming, copy it to the gateway.

Here is an example of the MNP url:

https://s1.bichara.com.br:8181/chkporta.php?user=832700&pwd=sdsfdg&tn=83881

66902. The 8388166902 is the outgoing phone number，The outgoing number in the

url should be replaced by a variable ${num}.

When configuring MNP url, it should be replaced with ${num}, Then paste:htt

ps://s1.bichara.com.br:8181/chkporta.php?user=832700&pwd=sdsfdg&tn=${num}.

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5.5 Routing Blacklist

Figure 5-5-1 Routing Blacklist Settings

Blacklists are used to add a phone number to the blacklist or to remove a phone

number from the blacklist. After opening the blacklist routing, in the incoming mode,

if the calling number is in the blacklist, it will be routed to hangup() to end the call.

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6. SMS

6.1 General

You can choose enable SMS Received, SMS Local Stored and SMS Status Report or

not.
Figure 6-1-1 SMS Settings

Sender Options
You can change sender options here, include resend and the time of resend.

Figure 6-1-2 Sender Options

Table 6-1-1 Description of Sender Options

Options Definition

The times that you will attempt to resend your failed

Resend Failed Message
message.

Repeat Same Message The times that you will resend the same message.

Verbose Verbose level of sending message.

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SMS to Email
This is a tool that makes it available for you to email account to transmit th

e SMS to other email boxes. The following settings realize that received SMS

through [email protected] transmit to [email protected], open

[email protected] and [email protected].

Figure 6-1-3 SMS to Email

Table 6-1-2 Types of E-mail Box

SMTP Security
E-mail Box Type SMTP Server SMTP Port
Connectivity

Gmail smtp.gmail.com 587 √

Hotmail smtp.live.com 587 √

Yahoo! smtp.mail.yahoo.co.in 587 ×

e-mail smtp.163.com 25 ×

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Table6-1-3 Definition of SMS to E-mail

Options Definition

When you switch on, the following options are available,

Enable
otherwise, the following options are unavailable.

Email Address To set the email address of an available email account. For

of Sender example:[email protected].

Domain To set outgoing mail server. For example: smtp.gmail.com.

To set the port number of outgoing mail server. (Default is

SMTP Port
25)

The login name of your existing email account. The login

SMTP User name of your existing email account. This option might be

Name different from your email address. Some email client doesn't

need the email postfix.

SMTP Password The password to login your existing email.

When you choose Yahoo and 163 free e-mails, this option is
TLS Enable
not available.

SMTP Server To set outgoing mail server. For example：mail.openvox.cn.

Destination Email
The first email address to receive the inbox message.
Address1

Destination Email
The second email address to receive the inbox message.
Address2

Destination Email The third email address to receive the inbox message.

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Address3

SMS Control
It allows the endpoint to send some specific KEY WORDS and corresponding

PASSWORD in response to operate the gateway. SMS is case-sensitive. In default, this

function is disabled.

Figure 6-1-4 SMS Control

For example, SMS control password is 123456 which has nothing to do with the login

password, you can send “get info 123456” to the module’s phone number to get

your gateway’s IP information.

Table 6-1-4 Definition of SMS Control

Options Definition

Enable ON (enable), OFF (disable).

The password to confirm that SMS makes the gateway

Password rebooted, shut down, restored configuration files and get info

on this gateway.

For example, the message formats:

reboot system PASSWORD：To reboot your whole gateway

SMS Format
（The PASSWORD is referring to the PASSWORD you set up

from option “PASSWORD” above.）

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Reboot asterisk PASSWORD：To restart your gateway core.

restore config PASSWORD：To reset the configuration files back

to the default factory settings.

Get info PASSWORD：To get your gateway IP address.

Switch on: When the size of the SMS inbox record file reaches

the max size, the system will cut a half of the file. New record
SMS inbox
will be retained.
Auto clean
Switch off: SMS record will remain, and the file size will

increase gradually. Default on, max size = 20 MB.

HTTP to SMS
Figure 6-1-5 HTTP to SMS Settings

Table 6-1-5 Definition of HTTP to SMS

Options Definition

Enable ON (enable), OFF (disable).

Enable CORS ON (enable), OFF (disable).

Allow Access Allow source domain names or IPs that can be accessed

Origin Domain across domains.

URL The URL for send sms.

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Username: the login username for send sms.

Password: the login password for send sms.

Phone number: the destination telephone number.

Message: the SMS contents.

Port: the gsm port for send sms. For example: gsm-1.1,

gsm-1.2.

Report: the sending result report format.

Timeout: how long to wait.

SMS to HTTP
Figure 6-1-6 SMS to HTTP Settings

Table 6-1-6 Definition of SMS to HTTP

Options Definition Example

host HTTP server IP address or domain. 172.16.6.171

port HTTP server port. 80

File path for receiving and processing the SMS

param1 smsreceive.php
data requests.

The name of the parameter used to process

param2 port
the 'port' value.

The name of the parameter used to process

param3 port name
the 'port name' value.

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The name of the parameter used to process

param4 message
the ' message ' value.

The name of the parameter used to process

param5 time
the ' time ' value.

6.2 SMS Sender

Click the box to select the corresponding port number, fill in the target number and

SMS content .Then you can send it. Different numbers should be separated by

symbols: '\r', '\n', space character, semicolon and comma. After sending a text

message, if the SMS status report is enabled, a detailed report of the SMS

transmission will be displayed below.

Figure 6-2-1 SMS Sender

6.3 SMS Inbox

On this page, you are allowed to scan, delete, clean up, and export each port’s

received SMS. Also, you are allowed to check messages by port, phone number, time

order and message keywords.

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Figure 6-3-1 SMS Inbox

6.4 SMS Outbox

On this page, you are allowed to scan, delete, clean up, and export each port’s

received SMS. Also, you are allowed to check messages by port, phone number, time

order and message keywords.

Figure 6-4-1 SMS Outbox

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6.5 SMS Forwarding

Using this function, you can forward incoming sms to your mobile. You can click

button to add new routing. For example：

Figure 6-5-1 SMS Forwarding

SMS received by lte-1.1 and lte-1.2, will be transferred to phone number

13923704563 through lte-1.3 or lte-1.4.

Figure 6-5-2 Create a Routing

For "Ascending" Policy, if you choose 2 or more ports members, it will use first

available port to transfer sms. For this case, if lte-1.3 is available, it will always use

lte-1.3 to transfer sms; otherwise, it will use lte-1.4 to transfer sms.

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7. Network

7.1 LAN Settings

There are three types of LAN port IP: Factory, Static, and DHCP. The Factory is the

default type with an IP of 172.16.98.1. This page is not editable when the LAN IPv4

type is selected as "Factory".

DNS Servers: A list of DNS IP address. Basically this info is from your local network

service provider.

A reserved IP address to access in case your gateway IP is not available. Remember to

set a similar network segment with the following address of your local PC.

Figure 7-1-1 LAN Settings

Table 7-1-1 Definition of LAN Settings

Options Definition

Interface The name of the network interface.

The method to get IP:

Type
Factory: Getting IP address by Slot Number (System

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information to check slot number).

Static: Manually set up your gateway IP.

DHCP: Automatically get IP from your local LAN.

MAC Address Physical address of your network interface.

IP Address The IP address of your gateway.

Netmask The subnet mask of your gateway.

Default Gateway Default gateway IP address.

7.2 WAN Settings

There are three types of WAN port IP, Disable, Static and DHCP. DHCP is the default

type. When you Choose IPv4 type is “Disable” or “DCHP”, this page is not editable.

Figure 7-2-1 WAN Settings

7.3 VPN Settings

The wireless gateway provides PPTP and N2N VPN connections, which allow users to

establish virtual private networks, encrypt communications, and achieve remote

access.

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Figure 7-3-1 PPTP VPN Settings

Figure 7-3-2 N2N VPN Settings

7.4 DDNS Settings

You can enable or disable DDNS (dynamic domain name server).

Figure 7-4-1 DDNS Settings

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Table 7-4-1 Definition of DDNS Settings

Options Definition

DDNS Enable/Disable DDNS (dynamic domain name server).

Type Set the type of DDNS server.

Username Your DDNS account’s login name.

Password Your DDNS account’s password.

Your domain The domain to which your web server will belong.

7.5 Toolkit

Some tools for checking network connections are provided, Ping commands and

route tracking are supported in the web user interface. You can set the

source/destination host address, port, and protocol to capture network packets.

Figure 7-5-1 Toolkit

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7.6 Firewall Settings

Figure 7-6-1 Firewall Settings

Table 7-6-1 Definition of Firewall Settings

Options Definition

If you want to use White/Black List, and security rules, you

Firewall Enable
must enable this option.

Whether to enable the Ping function. If the status is OFF:

Ping Enable
disable ping, the gateway does not allow ping.

Figure 7-6-2 White/Black List Settings

Table 7-6-2 Definition of White/Black List Settings

Options Definition

White/Black List Enable To enable White/Black list or not.

List IP IP is separated only by "," character.

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7.7 Security Rules

Figure 7-7-1 Security Rules Settings

Click "submit" button to submit and apply configuration.

If "List IP Settings" has no problem, you will see popup window like below. Please

read the warning and tips carefully. And Click "Apply" button in 1 minute. If time runs

out, this window will close automatically.

Figure 7-7-2 Security Rules Apply

If you see the windows like below. It means your configuration has been applied

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successfully.

Figure 7-7-3 Security Rules Apply

7.8 SIP Capture

You can capture the SIP packets on this page to facilitate location problems.

Figure 7-8-1 SIP Capture

Table 7-8-1 SIP Capture Settings

Options Definition

Interface You can choose eth0, eth1 or eth0:0.

Method-filter You can choose INVITE, OPTIONS, REGISTER or ALL.

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8. Advances

8.1 Asterisk API

When you make “Enable” switch to “ON”, this page is available.

Figure 8-1-1 Asterisk API

Table 8-1-1 Definition of Asterisk API

Options Definition

Port Network port number.

Manager Name Name of the manager without space.

Password for the manager. Allowed characters:

Manager secret
“-_+.<>&0-9a-zA-Z”. Length: 4-32 characters.

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If you want to deny some hosts or networks, you can

Deny use char & as separator. For example: 192.168.1.0/255.2

55.255.0&10.0.0.0/255.0.0.0.

If you want to permit some hosts or networks, you can

Permit use char & as separator. For example: 192.168.1.0/255.

255.255.0&10.0.0.0/255.0.0.0.

General information about the system and ability to run

System system management commands, such as Shutdown, Res

tart, and Reload.

The information about channels and the setting informa

Call
tion of the channel in use.

Log Logging information.

Verbose Verbose information.

Command Permission to run CLI commands. Write-only.

The information about queues and agents and the abilit

Agent
y to add queue members to a queue.

User Permission to send and receive User Event.

Config Ability to read and write configuration files.

DTMF Receive DTMF events. Read-only.

Reporting The ability to get information about the system.

CDR Call records. Read-only.

Dialplan Receive NewExten and Varset events. Read-only.

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Originate Permission to initiate new calls. Write-only.

All Select all or deselect all.

Please refer to the following figure to access the gateway API by telnet. 172.16.179.1

is the gateway’s IP, and 5038 is its API port.

Figure 8-1-2 Telnet Access Gateway API

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8.2 Asterisk CLI

On this page, you are allowed to run Asterisk commands.

Figure 8-2-1 Asterisk CLI

Command: Type your Asterisk CLI commands here to check or debug your gateway.

Notice: If you type “help” or “?” and execute it, the page will show you the shell

command.

8.3 Asterisk File Editor

On this page, you are allowed to edit and create configuration files. Click the file to

edit.

Figure 8-3-1 Asterisk File Editor

Click to create a new configuration file. After editing or

creating, you should reload Asterisk.

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8.4 Internet

In order to meet the requirements of some operators Package consumption flow, we

provide Internet access which can be set in batches (This function is only available on

the LTE module currently. Other hardware does not support the Internet function at

this time).

Figure 8-4-1 Asterisk Internet

8.5 Cloud Management

SWG and VoxStack series gateways both support OpenVox Cloud Management.
Figure 8-5-1 OpenVox Cloud Management

If your device is connected to the cloud management, the SSH and the web pages of

the gateway can be accessed through the cloud management, and it can be

monitored whether the device is connected to the cloud management platform. On

the cloud management platform, you can also count your device model, quantity,

distribution area, and so on which can provide you with efficient and excellent

service and experience.

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Table 8-5-1 Definition of Cloud Management

Options Definition

Enable Cloud
Turn on/off the cloud management.
Service

Currently supports two servers, one is China and the other is

Choose Service
the United States.

Registered account or email on the cloud management

Account
platform.

The password of the account registered on the cloud

Password
management platform.

Connection Whether currently connected to the cloud management

Status platform or not.

8.6 Balance

The wireless gateway provides a balance inquiry function for users who use the SIM

card in batches, it can automatically send a short message to the operator to query

the current balance on the SIM card, which is convenient, fast and accurate.

Figure 8-6-1 Balance

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Figure 8-6-2 Balance Settings

8.7 Phone Number

The wireless gateway also provides a number inquiry function, which can

automatically send a short message to the operator to query the number of the SIM

card that you use currently.

Figure 8-7-1 Phone Number

Figure 8-7-2 Phone Number

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9. Logs
On the “Log Settings” page, you should set the related logs on to scan the responding

logs page. For example, set “System Logs” on like the following, then you can turn to

“System” page for system logs. Otherwise, system logs are unavailable. Other log

pages are the same.

Figure 9-1 Logs Settings

Table 9-1 Definition of Logs

Options Definition

System Logs Whether to enable the system logs or not

Switch on: when the size of log file reaches the max size, the

system will cut a half of the file and the new logs will be

Auto clean retained;

(System Logs) Switch off: logs will remain, and the file size will increase

gradually.

The default is on, the default maxsize is 1MB.

SIP Logs Whether to open the sip logs or not

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Switch on: when the size of log file reaches the max size, the

system will cut a half of the file and the new logs will be

Auto clean retained;

(SIP logs) Switch off: logs will remain, and the file size will increase

gradually.

The default is on, the default maxsize is 100KB.

IAX Logs Whether to open the IAX logs or not

Switch on: when the size of log file reaches the max size, the

system will cut a half of the file and the new logs will be

Auto clean retained;

( IAX logs) Switch off: logs will remain, and the file size will increase

gradually.

The default is on, the default maxsize is 100KB.

Call Detail
Show Call Detail Records for each channel.
Record

Switch on: IMEI will be appended to the CDR gsm channel in

'From' or 'To'.
Append IMEI
Switch off: No appended IMEI.

The default is off.

Switch on: when the size of log file reaches the max size, the

system will cut a half of the file and the new logs will be
Auto clean
retained;
(Call Detail
Switch off: logs will remain, and the file size will increase
Record)
gradually.

The default is on, the default maxsize is 10MB.

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With the Syslog software, the gateway's logs and CDRs can be monitored and stored

locally on the PC.

Figure 9-2 Logs Settings

Figure 9-3 System Logs

You can easily browse your CDRs on the Web GUI, you also can delete, clean or

export your CDR information.

Figure 9-4 CDR

In the new version we have enriched the LOGS display which you can see each port

of the call outbound clearly.

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Figure 9-5 GSM Outbound

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