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                  OpenVox Communication Co Ltd


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                      SWG-20XX Gateway User Manual

                                        Version 1.0

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                                             OpenVox Communication Co Ltd




Address:10/F, Building 6-A, Baoneng Science and Technology Industrial Park, Longhua New District, Shenzhen, Guangdong,China  518109
Tel: +86-755-66630978, 82535461, 82535362
Business Contact:

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sales@openvox.cn
Technical Support: support@openvox.cn
Business Hours: 09:00-18:00(GMT+8) from Monday to Friday
URL: www.openvox.cn


                                            Thank You for Choosing OpenVox Products!


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.

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Trademarks
All other trademarks mentioned in this document are the property of their respective owners.

Revise History

Version

Release Date

Description

1.0

10/5/2018

Full text



Contents

1. Overview
1.1 What is SWG-20XX?
1.2 Product Introduction
1.3 Application
1.3.1 LCD And Buttons
1.3.2 Multifunction button
1.3.3 Console
1.4 Main Features
1.5 Physical Information
1.6 Software
2. System
2.1 Status
2.2 Time
2.3 Login Settings
2.4 General
2.4.1 Language Settings
2.4.2 Scheduled Reboot
2.5 Tools and Information
2.5.1 Reboot Tools
2.5.2 Update Firmware
2.5.3 Upload and Backup Configuration
2.5.4 Restore Configuration
2.6 Information
3. MODULE
3.1 MODULE Settings
3.1.1 Call Duration Limit Settings
3.2 DTMF
3.3 Toolkit
4. VOIP
4.1 VOIP Endpoints
4.1.1 Add New SIP Endpoint
4.1.2 Add New IAX2 Endpoint
4.2 Batch SIP Endpoints
4.3 Advanced SIP Settings
4.3.1 Networking
4.3.2 Paesing and Compatibility
4.3.3 Security
4.3.4 Media
4.3.5 Codec Settings
4.4 Advanced IAX2 Settings
4.4.1 General Settings
4.4.2 Music on Hold
4.4.3 Instruction of Codec Settings
4.4.4 Jitter Buffer Settings
4.4.5 Misc Settings
4.4.6 Quality of Service
5. Routing
5.1 Groups
5.2 Batch Creating rules
5.3 MNP Settings
6. SMS
6.1 General
6.1.1 Sender Options
6.1.2 SMS to Email
6.1.3 SMS Control
6.1.4 HTTP to SMS
6.1.5 SMS to HTTP
6.2 SMS Sender
6.3 SMS Inbox
6.4 SMS Outbox
6.5 SMS Forwarding
7. Network
7.1 LAN Settings
7.2 WAN Settings
7.3 VPN Settings
7.4 DDNS Settings
7.5 Toolkit
7.5.1 Ping and Traceroute
7.5.2 TCP Capture
7.6 Security Settings
7.6.1 Firewall Settings
7.6.2 White/Black List Settings
7.7 Security Rules
7.8 SIP Capture
8. Advances
8.1 Asterisk API
8.2 Asterisk CLI
8.3 Asterisk File Editor
8.3 Cloud Management
9. Logs
Appendix Feature List
General Info
VOIP Characters
Network
System Features
Management

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The SWG-20XX series gateways are available in a variety of models, and each model supports a different number of ports and frequency bands. The following table shows:

Model

Module

Ports

Network Interface

Band

USB

TF

Console

SWG-2016C

CDMA

16

2

800MHz@CDMA 2000

1

1

1

SWG-2016G

GSM

16

2

850/900/1800/1900MHz@GSM

1

1

1

SWG-2016L

LTE

16

2

LTE FDD: B1/B3/B5/B8
LTE TDD: B38/B38/B40/B41
WCDMA: B1/B8
TD-SCDMA: B34/B39
CDMA: BC0
GSM: 900/1800MHz

1

1

1

SWG-2032C

CDMA

32

2

800MHz@CDMA 2000

1

1

1

SWG-2032G

GSM

32

2

850/900/1800/1900MHz@GSM

1

1

1

SWG-2032L

LTE

32

2

LTE FDD: B1/B3/B5/B8
LTE TDD: B38/B38/B40/B41
WCDMA: B1/B8
TD-SCDMA: B34/B39
CDMA: BC0
GSM: 900/1800MHz

1

1

1


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1.3 Application

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1.3.1 LCD And Buttons

LED Indicator/Icon/Buttons

Color/ Icon

Staus

Display Icon











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Module Initiating,Disable

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No SIM Card

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Searching for Signal

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One grid Signal

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Two grid Signal

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Three grid Signal

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four grid Signal

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fives grid Signal

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Worst Signal Quality During a Call

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Medium Signal Quality During a Call

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Best Signal Quality During a Call

Network Status LED

Green and Flash

Network Connected

Power LED

Always Green

Power on

PER Button


OFF

Power down

ON

Power on

RST Button

Press and hold the RST button for 3-5 seconds. The display jumps to the "System Booting" page to restart the system.

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1.3.2 Multifunction button

  1. [▲] : Press this key to flip up
  2. OK :

Press this key in the signal interface - enter the menu

Press this key in the menu interface – Confrim

Press this key if it is Non-signal interface and there is no return option in the current interface - Back

      3. [▼] : Press this key to flip down

      4. Press any key in the signal interface to enter the menu interface.

      5. If no button is operated within 20S, return to the main interface.


The main factions are as follows:

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On the "Status" page, you will find all Modules, SIP, IAX2, Routing and Network information.
Figure 2-1 Systm Status




Table 2-1 Description of System Status

Options

Definition

Port

Number of each ports. 

Signal

Display the signal strength of in each channels of gateway. 

BER

Bit Error Rate. 

Carrier

Display the network carrier of current SIM card. 

Registration Status

Indicates the registration status of current module.

PDD

Post Dial Delay (PDD) is experienced by the originating customer as the time from the sending of the final dialed digit to the point at which they hear ring tone or other in-band information.Where the originating network is required to play an announcement before completing the call then this definition of PDD excludes the duration of such announcements. 

ACD

The Average Call Duration (ACD) is calculated by taking the sum of billable seconds (bill sec) of answered calls and dividing it by the number of these answered calls. 

ASR

Answer Seizure Ratio is a measure of network quality. Its calculated by taking the number of successfully answered calls and dividing by the total number of calls attempted. Since busy signals and other rejections by the called number count as call failures, the ASR value can vary depending on user behavior. ModuleStatus Show the status of port, include blank space and "READY". Black space means it is unavailable here and "Ready" means the port is available

Module Status

Display the status of the port. "Ready" means registering and "READY" means port is available

Remain Time

This value is multiplied by to step length is a rest call time.

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Table 2-2 Description of Time Settings

Options

Definition

System Time

Your gateway system time

Time Zone

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

POSIX TZ String

Posix time zone strings.

NTP Server 1

Time server domain or hostname. For example,
[time.asia.apple.com].

NTP Server 2

The first reserved NTP server. For example, [time.windows.com].

NTP Server 3

The second reserved NTP server. For example, [time.nist.gov].

Save Data

Save the Modify of the time settings

Sync from NTP

Sync time from NTP server. 

Sync from Client

Sync time from local machine.


For example, you can configure like this:
Figure 2-2 Time Settings

You can set your gateway time Sync from NTP or Sync from Client by pressing different buttons.

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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 user name 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 Description of Login Settings

Options

Definition

User Name

Define your username and password to manage your gateway
Allowed characters "-_+. < >&0-9a-zA-Z". Length: 1-32 characters.

Password

Allowed characters "-_+. < >&0-9a-zA-Z". Length: 4-32 characters.

Confirm Password

Please input the same password as 'Password' above.

Login Mode

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

Notice: Whenever you do some changes, do not forget to save your configuration.

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Figure 3-1 Module Settings

On this page, you can see your SIM Card information and module status,click action  button to configure the port.
Figure 3-2 Port Configuration

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


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 3-4 CLIR Application


Table 3-1 Definition of Module Settings

Options

Definition

Name

The alias of the each port. Input name without space here.
Allowed characters "-_+.<>&0-9a-zA-Z".Length: 1-32 characters. 

Speaker Volume

The speaker volume level, the range is 0-100.
This will adjust the loud speaker volume level by an AT command. 

Microphone Volume

The microphone volume, range is: 0-15.
This will change the microphone gain level by an AT command. 

Dial Prefix

The prefix number of outgoing calls from this channel 

PIN Code

 Personal identification numbers of SIM card. PIN code can be modified to prevent SIM card from being stolen. 

Custom AT commads when start

User custom AT commands when start system, use

""

 “|” to split AT command

CLIR

 Caller ID restriction, this function is used to hidden caller ID of SIM card number. The gateway will add '#31#' in front of mobile number. This function must support by Operator.

SMS Center Number

Your SMS center number of your local carrier. 

Module IMEI

Only CDMA module does not support modifying IMEI

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Now we can offer you two types of call duration limit, you can choose "Single Call Duration Limit" or "Call Duration Limitation" to control your calling time
Single Call Duration Limit: 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 by this port, it will limit your calling time within the product of
Step * Single Call Duration Limitation
And if your calling time overtops the value above, the system will hang up this call.
Figure 3-5 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 3-6 Call Duration Limitation Settings


The same algorithm with single time limitation, the total calling time of this port can't beyond the product of "Step" and "Call Duration Limitation".
If the duration of a call is less than "Minimum Charging Time", it will be not included in "Call Duration".
You can set a digit for "Alarm Threshold", when the call minutes less than this value, the gateway will send alarm info to designated phone.
You can enable your Auto Reset, then choose by day, by week, or by month.
Figure 3-7 Auto Reset Settings


Table 3-2 Description of Call Duration Limit Settings

Options

Definition

Step

Step length value range is 1-999s, step length multiplied by time of single call just said a single call duration time allowed. 

Enable Single Call Duration Limit

Definite maximum call duration for single call. Example: if Time of single call set to 10, the call will be disconnected after talking 10*step seconds.

Enable Call Duration Limitation

This function is to limit the total call duration of channel. The max call duration is between 1 to 999999 minutes. 

Minimum Charging Time

A single call over this time, Module side of the operators began to collect fees, unit for seconds. 

Alarm Threshold

 Define a threshold value of call minutes, while the call minutes less than this value, the gateway will send alarm information to designated phone.

Alarm Description

 Alarm port information description, which will be sent to user mobile phone with alarm information.

Alarm Phone Number

Receiving alarm phone number, user will received alarm message from gateway.

Enable Auto Reset

Automatic restore remaining talk time, that is, get total call minutes of each channel. 

Auto Reset Type

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

Next Reset Time

  Defined next reset date, system will count start from that date and work as Reset Period setting


You can save your configuration to other ports.
Figure 3-8 Save to Other Ports

If you have set like this, you will see many  on the Web GUI, you can set whether to check.


Notice: When you do some changes, you need to Save and Apply, then "Remain Time" will show as you set.
Your calling status will show on the main interface.


Figure 3-9 Module Information

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You can do some DTMF Detection Settings if you choose "MODULE –> DTMF".
Figure 3-10 DTMF Detection Settings


Notice: If you don't have special need, you don't have to modify these settings. You can just choose "Default".


Table 3-3 Description of DTMF Detection Settings

Options

Definition

DTMF Normal Twist and Reverse Twist

It is the difference in power between the row and column energies. Normal Twist is where the Column energy is greater than the Row energy. Reverse Twist is where the Row energy is greater.

DTMF Relative Peak Row

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

DTMF Relative Peak Col

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

DTMF Hits Begin

Sampling matching value. You can choose 2 or 3.

DTMF Misses End

The time interval between the two digits you input. Adjust the speed of input. The smaller value represents the shorter intervals.

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You can get USSD information, send AT command and check number with this module. When you have a debug of the module, AT command is useful.
Figure 3-11 Function Options


Table 3-4 Description of Definition of Functions

Options

Definition

Check Number

 Enter a known number (like your mobile phone) to check what number it is of the SIM card. Click "Execute", then 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.

Get USSD

Enter a specific USSD number (For example,*142# to check your SIM card's balance. This USSD number is might be different from different carriers) to get the USSD information. The gateway will try to get by AT commands. 

AT Command

To perform some specific AT commands. This is useful when you have a debug of the modem. e.g. perform [ AT+CSQ ] to check what signal qualify it is. In AT commands, there is no difference between "a" and "A"


If you want to send AT command, first you should input your command, then select certain ports and choose "Copy to Selected", finally choose "Execute".
Figure 3-12 AT Command Example

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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 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 4-2 None Registration


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 4-3 Server


Also you can choose registration by "This gateway registers with the endpoint", it's the same with "None", except name and password.
Figure 4-4 Client


Table 4-1 Definiton of SIP Options

Options

Definition

Name

Display name

Username

Register name in your SIP server

Password

Authenticating with the gateway and characters are allowed.

Registration

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;

Hostname or IP Address

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

Transport

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 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.

NAT Traversal

No --- Use Rport if the remote side says to use it. Force Rport on --- Force Rport to always be on. Yes --- Force Rport to always be on and perform comedia RTP handling. Rport if requested and comedia --- Use Rport if the remote side says to use it and perform comedia RTP handling.


Advanced——Registration Options
Figure 4-5 Advanced Registration Options


Table 4-2 Definition of Registration Options

Options

Definition

Authentication User

A username to use only for registration.

Register Extension

When Gateway registers as a SIP user agent to a SIP proxy (provider), calls from this provider connect to this local extension.

From User

A username to identify the gateway to this endpoint.

From Domain

A domain to identify the gateway to this endpoint.

Remote Secret

A password which is only used if the gateway registers to the remote side.

Port

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

Qualify

Whether or not to check the endpoint's connection status

Qualify Frequency

How often, in seconds, to check the endpoint's connection status.

Outbound Proxy

A proxy to which the gateway will send all outbound signalling instead of sending signalling dirrectly to endpoints.


Call Settings
Figure 4-6 Call Settings


Table 4-3 Definition of Call Options

Options

Definition

DTMF Mode

Set default DTMF Mode for sending DTMF. Default: rfc2833. Other options: 'info', SIP INFO message (application/dtmf-relay); 'Inband', Inband audio (require 64kbit codec -alaw, ulaw).

Trust Remote-Party-ID

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

Send Remote-Party-ID

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

Remote Party ID Format

How to set the Remote-Party-ID header: from Remote-Party-ID or from P-Asserted-Identity.

Caller ID Presentation

Whether or not to display Caller ID.

Call Limit

Usually used when this sip work as a trunk. To limit number of maximum channels supported by the sip trunk.


Advanced:——Signaling Settings
Figure 4-7 Signaling Settings


Table 4-4 Definition of Signaling Options

Options

Definition

Progress Inband

Whether there is ringing tone.
Never: Indicates that incoming calls are never applicable.
Optional values: yes / no / never. Default: yes

Append user=phone to URI

 Whether or not to Add 'user = phone' to UPIS to include a valid phone number in the URI. 

Add Q.850 Reason Headers

If it is available, Whether or not to add a reason header and use it.

Honor SDP Version

Whether or not to display Caller ID. 

Allow Transfers

Whether or not to globally enable transfers. Choosing 'no' will disable all transfers (unless enabled in peers or users). Default is enabled.

Allow Promiscuous Redirects

Whether or not to allow 302 or REDIR to non-local SIP address. Note that promiscredir when redirects are made to the local system will cause loops since this gateway is incapable of performing a "hairpin" call.

Max Forwards

 Setting for the SIP Max-Forwards header (loop prevention). Send TRYING on REGISTER Send a 100 Trying when the endpoint registers. 

Outbound Proxy

 A proxy to which the gateway will send all outbound signaling instead of sending signaling directly to endpoints.


Advanced——Timer Settings
Figure 4-8 Timer Settings


Table 4-5 Definition of Timer Options

Options

Definition

Default T1 Timer

This timer is used primarily in INVITE transactions. The default for Timer T1 is 500ms or the measured run-trip time between the gateway and the device if you have qualify=yes for the device.

Call Setup Timer

 If a provisional response is not received in this amount of time, the call will auto-congest. Defaults to 64 times the default T1 timer. 

Session Timers

Session-Timers feature operates in the following three modes: originate, Request and run session-timers always; accept, run session-timers only when requested by other UA; refuse, do not run session timers in any case.

Minimum Session

 Minimum session refresh interval in seconds. Default is 90secs.

Maximum Session Refresh Interval

Maximum session refresh interval in seconds. Defaults to 1800secs.

Session Refresher

The session refresher, uac or uas. Defaults to uas.

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You can click  button to add a new IAX2 endpoint, and if you want to modify existed endpoints, you can click   button.
There are 3 kinds of registration types for choose. You can choose None, Endpoint registers with this gateway(work as a Server) or This gateway registers with the endpoint(work as a Client).
You can configure as follows:
If you set up a IAx2 endpoint by registration "None" to a server, then you can't register other IAX2 endpoints to this server, just authenticate the username and password.


Figure 4-9 None Registrarion


For convenience, we have designed a method that you can register your IAX2 endpoint to your gateway, thus your gateway just work as a server.
Figure 4-10 Server


Also you can choose registration by "This gateway registers with the endpoint", it will work as a Client.
Figure 4-11 Client


Table 4-6 Definition of IAX2 Options

Options

Definition

Name

Display name

Username

Authenication name in your IAX2 server

Password

Authenticating with the gateway and characters are allowed.

Registration

None --- Not registering;   

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

Hostname or
IP Address

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

Auth

There are three authentication methods that are supported: md5, plaintext and rsa. The least secure is "plaintext", which sends passwords cleartext across the net. "md5" uses a challenge/response md5 sum arrangement, but still requires both ends have plain text access to the secret. "rsa" allows unidirectional secret knowledge through public/private keys.If "rsa" authentication is used, "inkeys" is a list of acceptable public keys on the local system that can be used to authenticate the remote peer, separated by the ":" character. "outkey" is a single, private key to use to authenticate to the other side.

Transfer

 This application allows you to transfer calls.

Trunk

"trunk=yes" Purpose: To obtain a better chart of actual bandwidth usage per codec as seen "on-the-wire" when using IAX2 trunking between two Asterisk telephony servers.


Advanced——Registration Options
Figure 4-12 Registration Options


Table 4-7 Definition of Registration Options

Options

Definition

Qualify, Qualify Freq Ok, Qualify Freq Not Ok

The qualify, qualifyfreqok and qualifyfreqnotok settings are used to determine the status availability of an IAX peer. If a peer is consdered to be in a reachable (OK or LAGGED) state, it is queried for availability every "qualifyfreqok" milliseconds. If it is considered to be in an UNREACHABLE state, it is queried for availability every "qualifyfreqnotok" milliseconds.The qualify= setting turns the qualify system on (if the "yes" or xxx options are used) or off (if qualify=no, which is by default). The millisecond value of the qualify= setting specifies the maximum response time of the availability acknowledgement before the peer is considered to be in a "LAGGED" state.

Qualify Smothing

Use an average of the last two PONG result to reduce falsely detected LAGGED host. The default is 'no'.

Port

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


IAX2 Encryption
Figure 4-13 IAX2 Encryption


Table 4-8 Definition of Encrytion Options

Options

Definition

Encryption

Enable IAX2 encryption. The default is no. 

Force Encryption

Force encryption insures no connection is established unless both sides support encryption. By turning this option on, encryption is automatically; turned on as well. The default is no


IAX2 Trunk Settings
Figure 4-14 IAX2Trunk Settings


Table 4-9 Definition of Trunk Options

Options

Definition

Trunk Max Size

 Defaults to 128000 bytes, which supports up to 800; calls of ulaw at 20ms a frame. 

Trunk MTU

With a large amount of traffic on IAX2 trunk, there is a risk of bad voice quality when allowing the Linux system to handle fragmentation of UDP packets. Depending on the side of each payload, allowing the OS to handle fragmentation may not be very efficient. This setting sets the maximum transmission unit for AIX2 UDP trunking. The default is 1240 bytes which means if a trunk's payload is over 1240 bytes for every 20ms it will be broken into multiple 1240 bytes messages. Zero disables this functionality and let's the OS handle fragmentation.

Trunk Frequency

How frequently to send trunk msgs (in ms). This is 20ms by default. 

Trunk Time Stamps

Should we send timestamps for the individual sub_frames within trunk frames? There is a small bandwith use for these (less than 1kbps/call), but they ensure that frame timestamps get sent end-to-end properly. If both ends of all your trunks go directly to TDM, _and_your trunkfreq equals the frame length for your codecs, you can probably suppress these. The receiver must also need to have it enabled.

Min. RegExpire

Minimum amounts of time that IAX2 peers can request as a registration interval (in seconds).

Max. RegExpire

Maximum amounts of time that IAX2 peers can request as a registration expiration interval(in seconds). 

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4.2 Batch SIP Endpoints

In this page,you can generate multiple SIP Extentations at the same time.


Figure 4-15 Multiple SIP Extentations Settings


You can fill in the user name, password, domain name or IP address, port, and registration mode on the firt line and select the number of SIPs to be created. You can create up to the same number of SIP endpoints as the number of device ports at a time. After the above configuration, click Batch Setup and save it to create SIP endpoints in batches.
Table 4-10 Definition of Multiple SIP Extentations

Options

Definition

Name

Display name

Username

Register name in your SIP server

Password

Authenticating with the gateway and characters are allowed.

Registration

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;

Hostname or IP Address

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

AutoPassword

Tick - Automatically increments based on the password entered in the first lineDo not check - All SIP endpoints have the same password as the first one.


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4.3 Advanced SIP Settings

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Networking General
Figure 4-16 Networking General


Table 4-11 Definition of Networking General Optiongs

Options

Definition

UDP Bind Port

UDP Bind Port

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 Authentication Timeout

The maximum number of seconds a client has to authenticate. If the client does not authenticate before this timeout expires, the client will be disconnected.(default value is: 30 seconds).

TCP Authentication Limit

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

Enable Hostname Lookup

Enable DNS SRV lookups on outbound calls Note: the gateway only uses the first host in SRV records Disabling DNS SRV lookups disables the ability to place SIP calls based on domain names to some other SIP users on the Internet specifying a port in a SIP peer definition or when dialing outbound calls with suppress SRV lookups for that peer or call.

Enable Internal SIP Call

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

Internal SIP Call Prefix

Specify a prefix before routing the internal calls.


NAT Settings
Figure 4-17 NAT Settings


Table 4-12 Definition of NAT Settings Options

Options

Definition

Local Network

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 NATed 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 List

Local IP address list that you added.

Subscribe Network Change Event

Through the use of the test_stun_monitor module, the gateway has the ability to detect when the perceived external network address has changed. When the stun_monitor is installed and 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 only takes effect once res_stun_monitor is configured. If res_stun_monitor is enabled and you wish to not generate all outbound registrations on a network change, use the option below to disable this feature.

Match External Address Locally

Only substitute the externaddr or externhost setting if it matches.

Dynamic Exclude Static

Disallow all dynamic hosts from registering as any IP address used for statically defined hosts. This helps avoid the configuration error of allowing your users to register at the same address as a SIP provider.

Externally Mapped TCP Port

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

External Hostname

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

Hostname Refresh Interval

How often to perform a hostname lookup. This can be useful when your NAT device lets you choose the port mapping, but the IP address is dynamic. Beware, you might suffer from service disruption when the name server resolution fails.


RTP Settings
Figure 4-18 RTP Settings


Table 4-13 Definition of RTP Settings Options

Options

Definition

Start of RTP Port Range

 Start of range of port numbers to be used for RTP

End of RTP port Range

End of port numbers to be used for RTP

RTPTimeout

RTP Timeout retransmission time

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Figure 4-19 Paesing and Compatibility


Table 4-14 Instruction of Parsing and Compatibility

Options

Definition

Strict RFC Interpretation

Check header tags, character conversion in URIs, and multiline headers for strict SIP compatibility(default is yes)

Send Compact Headers

Send compact SIP headers 

SDP Owner

Allows you to change the username filed in the SDP owner string. This filed MUST NOT contain spaces. 

Disallowed SIP Methods

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

Shrink Caller ID

The shrinkcallerid function removes '(', ' ', ')', non-trailing '.', and '-' not in square brackets. For example, the caller id value 555.5555 becomes 5555555 when this option is enabled. Disabling this option results in no modification of the caller id value, which is necessary when the caller id represents something that must be preserved. By default this option is on.

Maximum Registration Expiry

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

Minimum Registration Expiry

Minimum length of registrations/subscriptions (default 60).

Default Registration Expiry

Default length of incoming/outgoing registration.

Registration Timeout

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

Number of Registration

Attempts Enter '0' for unlimited Number of registration attempts before we give up. 0 = continue forever, hammering the other server until it accepts the registration. Default is 0 tries, continue forever.

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Figure 4-20 Security Settings


Table 4-15 Instruction of Security

Options

Definition

Match Auth Username

If available, match user entry using the 'username' field from the authentication line instead of the 'from' field.

Realm

Realm for digest authentication. Realms MUST be globally unique according to RFC 3261. Set this to your host name or domain name.

Use Domain as Realm

Use the domain from the SIP Domains setting as the realm. In this case, the realm will be based on the request 'to' or 'from' header and should match one of the domain. Otherwise, the configured 'realm' value will be used.

Always Auth Reject

When an incoming INVITE or REGISTER is to be rejected, for any reason, always reject with an identical response equivalent to valid username and invalid password/hash instead of letting the requester know whether there was a matching user or peer for their request. This reduces the ability of an attacker to scan for valid SIP usernames. This option is set to 'yes' by default.

Authenticate Options Requests

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

Allow Guest Calling

Allow or reject guest calls (default is yes, to allow). If your gateway is connected to the Internet and you allow guest calls, you want to check which services you offer everyone out there, by enabling them in the default context.

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Figure 4-22 Media Settings


Table 4-16 Instruction of Media

Options

Definition

Premature Media

Some ISDN links send empty media frames before the call is in ringing or progress state. The SIP channel will then send 183 indicating early media which will be empty - thus users get no ring signal. Setting this to "yes" will stop any media before we have call progress (meaning the SIP channel will not send 183 Session Progress for early media). Default is 'yes'. Also make sure that the SIP peer is configured with progressinband=never. In order for 'noanswer' applications to work, you need to run the progress() application in the priority before the app.

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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Figure 4-23 General Settings


Table 4-17 Instruction of General

Options

Definition

Bind Port

Bind port and bindaddr may be specified

Enable IAXCompat

More than once to bind to multiple addresses, but the first will be the default.

Enable Nochecksums

Set iaxcompat to yes if you plan to use layered switches or some other scenario which may cause some delay when doing a lookup in the dialplan. It incurs a small performance hit to enable it. This option cause Asterisk to spawn a separate thread when it receives an IAX DPREQ (Dialplan Request) instead of blocking while it waits for a response.

Enable Delay Reject

Disable UDP checksums (if no checksums is set, then no checksums will be calculated/checked on system supporting the feature)

ADSI

ADSI (Analog Display Services Interface) can be enable if you have (or may have) ADSI compatible CPE equipment.

SRV Loopup

Whether or not to perform an SRV lookup on outbound calls

AMA Flags

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

autokill

If we don't get ACK to our NEW within 2000ms,and autokill is set to yes, then we cancel the whole thing(that's enough time for one retransmission only ).This is used to keep things from stalling for a long time for a host that is not available for bad connections.

Language

You may specify a global default language for users. This can be specified also on a per-user basis. If omitted, will fallback to English(en)

Account Code

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

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Figure 4-24 Music on Hold Settings


Table 4-18 Instruction of Music on Hold

Options

Definition

Mohsuggest

The 'Mohsuggest' option specifies which music on hold class to suggest to the peer channel when this channel place the peer on hold. It may be specified globally or on a per-user or per-peer basis.

Mohinterpret

You may specify a global default language for users. This can be specified also on a per-user basis. If omitted, will fall back to English(en)

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Figure 4-25 Codec Settings


Table 4-19 Instruction of Codec Settings

Options

Definition

Band Width

Specify bandwith of low, medium, or high to control which codes are used in general

Disallow

Fine tune codes here using "allow" and "disallow" clause with specific codes

Allow

Fine tune codes here using "allow" and "disallow" clause with specific codes

Codec Priority

Codec priority controls the codec negotiation of an inbound IAX2 call. This option is inherited to all user entity separately which will override the setting in general.

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Figure 4-26 Jitter Buffer


Table 4-20 Instruction of Jitter Buffer

Options

Definition

Jitter Buffer

Global default as to whether you want the jitter buffer at all

Force Jitter Buffer

In the ideal world, when we bridge VoIP channels we don't want to jitter buffering on the switch, since the endpoints 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.

Max Jitter Buffers

A maximum size for the jitter buffer

Resyncthreshold

When the jitter buffer notice a significant change in delay that continue over a few frames, it will resync, assuming that the change in delay was caused by a timestamping mix-up. The threshold for noticing a change in delay is measured as twice the measured jitter plus this resync threshold.

Max Jitter Interps

The maximum number of interpolation frames the jitter buffer should return in a row. Since some clients do not send CNG/DTX frames to indicate silence, the jitter buffer will assume silence has begun after returning this many interpolations. This prevents interpolating throughout a long silence.

Jitter Target Extra

Number of milliseconds by which the new jitter buffer will pad its size. The default is 40, so without modification, the new jitter buffer will set its size to the jitter value may help if your network normally has low jitter, but occasionally has spikes.

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4.4.5 Misc Settings

Figure 4-27 Misc Settings


Table 4-21 Instruction of Misc Settings

Options

Definition

IAX Thread Count

Establishes the number of iax helper thread to handle I/O

IAX Max Thread Count

Establishes the number of extra dynamic threads that may by spawned to handle I/O

Max Call Number

The 'maxcallnumbers' 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 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.

MaxCallNumbers_Nonvalidated

The 'maxcallnumbers-nonvalidated' is used to set the combined number of call numbers that can be allocated for connections where call token validation has been disabled. Unlike the 'maxcallnumbers' option, this limit is not separate for each individual IP address. Any connection resulting in a non-call token validated call number being allocated contributes to this limit. For use cases, see the call should be sufficient in most cases.

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Figure 4-28 Quality of Service


Table 4-22 Instruction of Quality of Service

Options

Definition

Tos

Type of service

Cos

Class of service

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Figure 5-1 Routing Rules

You are allowed to set up new routing rule 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 Rule:
You can click  button to set up your routings.


Figure 5-2 Example of Set up Routing Rule



The figure above shows that all the phones in the group ALL are transferred to the SIP-1234 terminal.
Table 5-1 Definition of Routing Options

Options

Definition

Routing Name

The name of this route. Should be used to describe what types of calls this route matches (for example, 'SIP2CDMA' or 'CDAM2SIP').

Call Comes in From

The launching point of incoming calls.

Send Call Through

The destination to receive the incoming calls.


Table 5-2 Description of Advanced Routing Rule

Options

Definition

Dial Patterns that will use this Route

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).Rules: X matches any digit from 0-9 Z matches any digit from 1-9 N matches any digit from 2-9 [1237-9] matches any digit in the brackets (example: 1,2,3,7,8,9) . wildcard: matches one or more dialed digits. prepend: Digits to prepend to a successful matchIf 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 matchThe 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 Name to

What caller ID name would you like to set before sending this call to the endpoint.

Forward Number

What destination number will you dial? This is very useful when you have a transfer call.

Custom Context

User-defined dialing rules

Failover Call Through Number

The gateway will attempt to send the call out each of these in the order you specify. You can create various time routes and use these time conditions to limit some specific calls.


Figure 5-3 Time Patterns that will use this Route

If you configure like this, then from January to March, from the first day to the last day of these months, from Monday to Thursday, from 00:00 to 02:00, during this time (meet all above time conditions), all calls will follow this route. And the time will synchronize with your Sever time.


Figure 5-4 Failover Call Through Number

You can add one or more "Failover Call Through Numbers".

...

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.


Figure 5-5 Routing Group

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5.2 Batch Creating rules

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

Figure 5-6 Batch Creating rules Group


You can configure the SIM Number, SIP trunk and calling Number for each port.And then, click "save" to batch creating multiple Routing rules.By an attention, the SIP trunk must be configures and the SIM number and calling Number can be emply.
Table 5-3 Description of Advanced Routing Rule

Options

Definition

Forward Number

What destination number will you dial? This is very useful when you have a transfer call.

SIP Trunk

Inbound and outbound calls through designated SIP trunks

Set the Caller ID Name to

What caller ID name would you like to set before sending this call to the endpoint.


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5.3 MNP Settings

Mobile Number Portability allows switching between mobile phone operators without changing the mobile number. Sounds simple, but there are loads of tasks performed behind the scene at the operator end.
The URL is shown in the password string way. So please type the url in other place such a txt file, check it, then copy it to the gateway. The outgoing number in the url should be replaced by the variables ${num}.
Here is an example of the MNP url:
https://s1.bichara.com.br:8181/chkporta.php?user=832700&pwd=sdsfdg&tn=8388166902 
The 8388166902 is the outgoing phone number, when config the MNP url, should replce it with ${num}. Then it turns to https://s1.bichara.com.br:8181/chkporta.php?user=832700&pwd=sdsfdg&tn=${num}


Figrue 5-7 MNP Settings

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6. SMS

...

You can change sender options here, include resend, times of resend.
Figure 6-2 Sender Options


Table 6-1 Description of Sender Options

Options

Definition

Resend Failed Message

The times that you will attempt to resend your failed message. 

Repeat Same Message

The times that you will resend the same message. 

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6.1.2 SMS to Email

This is a tool that makes it available for you to email account to transmit the SMS to other email boxes. The following settings realize that received SMS through openvpnvoip@gmail.com transmit to openvpnvoip@yahoo.com.cn, openvpnvoip@hotmail.com and support@openvox.cn
Figure 6-3 SMS to Email


Table6-2 Types of E-mail Box

E-mail Box Type

SMTP Server

SMTP Port

SMTP Security Connectivity

Gmail

smtp.gmail.com

587

HotMail

smtp.live.com

587

Yahoo!

smtp.mail.yahoo.co.in

587

×

e-mail

smtp.163.com

25

×


Table6-3 Definition of SMS to E-mail

Options

Definition

Enable

When you choose on, the following options are available, otherwise, unavailable.

Email Address of Sender

To set the email address of an available email account. For example, openvpnvoip@gmail.com.

Domain

To set outgoing mail server. e.g. smtp.gmail.com

SMTP Port

To set port number of outgoing mail server. (Default is 25)

SMTP User Name

The login name of your existing email account. This option might be different from your email address. Some email client doesn't need the email postfix

SMTP Password

The password to login your existing email.

TLS Enable

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

SMTP Server

To set outgoing mail server. e.g. mail.openvox.cn.

Destination Email Address1

The first email address to receive the inbox message.

Destination Email Address2

The second email address to receive the inbox message.

Destination Email Address3

The third email address to receive the inbox message.

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Allowing endpoints to send some specific KEY WORDS and corresponding PASSWORD to operate the gateway and message is case-sensitive. In default, this function is disabled.

Figure 6-4 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 6-4 Definition of SMS Control

Options

Definition

Enable

ON(enable), OFF(disable)

Password

The password to confirm that SMS makes the gateway rebooted, shut down, restored configuration files and get info on this gateway.

SMS Format

For example, the message formats:
reboot system PASSWORD: To reboot your whole gateway.
The PASSWORD is referring to the PASSWORD you set up from option "PASSWORD" above.
Reboot asterisk PASSWORD: To restart your gateway core.
Restore configs PASSWORD: To reset the configuration files back to the default factory settings.
Get info PASSWORD: To get your gateway IP address

SMS inbox Auto clean

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 will be retained.
switch off: SMS record will remain, and the file size will increase gradually. default on, max size = 20 MB

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Using this feature, you can forward incoming sms to your mobile. You can click   button to add new routing. 
Such as:
Figure 6-10 SMS Forwarding Rules


SMS received by cdma-1.1 and cdma-1.2, cdma-1.4, will be transfered to phone number 18664565204 through port cdma-1.8 or cdma-1.10.
Figure 6-11 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 cdma-1.8 is availble, it will always use cdma-1.8 to trnasfer sms; Otherwise, it will use cdma-1.10 to transfer sms.

...

There are three types of LAN port IP, Factory, Static and DHCP. Factory is the default type, and it is 172.16.98.1. When you Choose LAN IPv4 type is "Factory", this page is not editable. 
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 LAN Settings


Table 7-1 Definition of LAN Settings

Options

Definition

Interface

The name of network interface.

Type

The method to get IP. 
Factory: Getting IP address by Slot Number
(System information to check slot number). 
Static: manually set up your gateway IP. 
DHCP: automatically get IP from your local LAN.

MAC

Physical address of your network interface.

Address

The IP address of your gateway.

Netmsk

The subnet mask of your gateway.

Default Gateway

Default getaway IP address.


DNS Servers:  A list of DNS IP address. Basically this info is from your local network service provider,and you can fill in four DNS servers.

...

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 WAN Settings


Table 7-2 Definition of WAN Settings

Options

Definition

Interface

The name of network interface.

Type

The method to get IP. 
Factory: Getting IP address by Slot Number
(System information to check slot number). 
Static: manually set up your gateway IP. 
DHCP: automatically get IP from your local LAN.

MAC

Physical address of your network interface.

Address

The IP address of your gateway.

Netmsk

The subnet mask of your gateway.

Default Gateway

Default getaway IP address.

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7.3 VPN Settings

SWG-20XX series gateways support PPTP VPN.
Figure 7-3 VPN Settings


Table 7-3 Definition of VPN Settings

Options

Definition

VPN Type

None – close VPN
PPTP VPN – use PPTP VPN

server

The server's IP address

Account

Server account

Password

The server's password

Use MPPE

Whether to use MPPE

Connection Status

Is it successful to connect to the server


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7.4 DDNS Settings

You can enable or disable DDNS (dynamic domain name server).
Figure 7-4 DDNS Settings


Table7-4 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.

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7.5 Toolkit

...

You can capture the tcp packets on the page to facilitate locationg problems.
Figure 7-6 TCP Capture


Table7-5 Definition of DDNS Settings

Options

Definition

Inferface

You can choose eth0 or eth1 

Source host

Source host IP

Destination host

Destination host IP

Port

Which port you want to capture?

Protocol

Which protocol you want to capture?


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7.6 Security Settings

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Figure 7-7 Firewall Settings


Table 7-6 Deginition of Firewall Settings

Options

Definition

Firewall Enale

If you want to use White/Black List, and security rules, you must enble this option. 

Ping Enable

To disable ping or not. OFF: disable ping. This gateway will not allow to ping. 

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White List Enbale: To enable white list or not. 
List IP Settings: IPs are separated only by "," character. 
Figure 7-8 White/Black List Settings


Click "Save" button to save configration; 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 7-9 Firewall Rules Apply


If you see windows like below. It means your configuration has been applied successfully.
Figure 7-10 Firewall Rules Apply

...

Figure 7-11 Security Rules


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 7-12 Security Rules Apply


If you see windows like below. It means your configuration has been applied successfully.
Figure 7-13 Security Rules Apply

...

You can capture the SIP packets on the page to facilitate locationg problems.
Figure 7-14 SIP Capture


Table 7-7 SIP Capture Settings

Options

Definition

Inferface

You can choose eth0 or eth1 

Method-filter

You can choose INVITE, OPTIONS and REGISTER


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8. Advances

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When you make "Enable" switch to "ON", this page is available.
Figure 8-1 Asterisk API


Table 8-1 Definition of Asterisk API

Options

Definition

Port

Network port number

Manager Name

Name of the manager without space

Manager secret

Password for the manager. Characters: Allowed characters "-_+.<>&0-9a-zA-Z". Length:4-32 characters.

Deny

 If you want to deny many hosts or networks, use char & as separator.Example: 0.0.0.0/0.0.0.0 or 
192.168.1.0/255.255.255.0&10.0.0.0/255.0.0.0

Permit

If you want to permit many hosts or network, use char & as separator. Example: 0.0.0.0/0.0.0.0 or
192.168.1.0/255.255.255.0&10.0.0.0/255.0.0.0

System

General information about the system and ability to run system management commands, <br/>such as Shutdown, Restart, and Reload.

Call

Information about channels and ability to set information in a running channel.

Log

Logging information. Read-only. (Defined but not yet used.)

Verbose

Verbose information. Read-only. (Defined but not yet used.)

Command

Permission to run CLI commands. Write-only.

Agent

Information about queues and agents and ability to add queue members to a queue.

User

Permission to send and receive UserEvent.

Config

Ability to read and write configuration files.

DTMF

Receive DTMF events. Read-only.

Reporting

Ability to get information about the system. CDR Output of cdr, manager, if loaded.

CDR

Call records. Read-only.

Dialplan

Receive NewExten and Varset events. Read-only.

Originate

Permission to originate new calls. Write-only.

All

Select all or deselect all.


Once you set like the above figure, the host 172.16.100.110/255.255.0.0 is allowed to access the gateway API. 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 8-2 Telnet Access Gateway API

...

In this page, you are allowed to run Asterisk commands.
Figure 8-3 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 executable commands.

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8.3 Asterisk File Editor

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SWG-20xx series gateways support OpenVox Cloud Management.
Figure 8-5 Cloud Management


If your device is connected to the cloud management, the SSH and 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.
Table 8-2 Definition of Cloud Management

Options

Definition

Enable Cloud Service

Turn on/off cloud management

Choose Service

Currently supports two servers, one is China and the other is the United States.

Account

Registered account or email on the cloud management platform

Password

The password of the account registered on the cloud management platform

Connection Status

Is it currently connected to the cloud management platform?


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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 is unavailable. And the same with other log pages.
Figure 9-1 Log Settings



Figure 9-2 System Logs


You can scan your CDR easily on web GUI, and also you can delete, clean up or export your CDR information.
Figure 9-3 CDR Output


Recently we have made our LOGS display richer, you can see your Outbound of every port clearly.
Figure 9-4 Outbound


Table9-1 definition of Logs

Options

Definition

System Logs

Whether enable or disable system log.

Auto clean
(System Logs)

switch on : when the size of log file reaches the max size, the system will cut a half of the file. New logs will be retained;    
switch off : logs will remain, and the file size will increase gradually. default on, maxsize=1M.

SIP Logs

Whether enable or disable SIP log.

Auto clean
 (SIP logs)

switch on: when the size of log file reaches the max size, the system will cut a half of the file. New logs will be retained.           
switch off: logs will remain, and the file size will increase gradually. default on, maxsize=100KB.

IAX Logs

Whether enable or disable IAX log.

Auto clean( IAX logs)

switch on: when the size of log file reaches the max size, the system will cut a half of the file. New logs will be retained.
switch off: logs will remain, and the file size will increase gradually. default on, maxsize=100KB.

Call Detail Record

Displaying Call Detail Records for each channel. 

Auto clean (CDR logs)

switch on : when the size of log file reaches the max size, the system will cut a half of the file. New logs will be retained.
 switch off : logs will remain, and the file size will increase gradually. default on, max size=20MB.

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Appendix Feature List

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