SWG-2016/32 User Manual
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OpenVox Communication Co. ,LTD. Website: www.openvox.cn
OpenVox Communication Co Ltd
SWG-2016/32 Gateway User Manual
Version 1.0
SWG-2016/32 User Manual
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OpenVox Communication Co. ,LTD. Website: www.openvox.cn
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: 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!
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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.
Trademarks
All other trademarks mentioned in this document are the property of their respective owners.
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Revise History
Version
Release Date
Description
1.0
10/5/2018
Full text
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Contents
1. Overview .......................................................................................................................................... 9
1.1 What is SWG-2016/32？ ........................................................................................................ 9
1.2 Product Introduction .............................................................................................................. 9
1.3 Application ........................................................................................................................... 10
1.3.1 LCD And Buttons ....................................................................................................... 10
1.3.2 Multifunction button ................................................................................................ 11
1.3.3 Console ...................................................................................................................... 14
1.4 Main Features ...................................................................................................................... 15
1.5 Physical Information ............................................................................................................. 15
1.6 Software ............................................................................................................................... 16
2. System ............................................................................................................................................ 16
2.1 Status .................................................................................................................................... 16
2.2 Time ...................................................................................................................................... 18
2.3 Login Settings ....................................................................................................................... 19
2.4 General ................................................................................................................................. 20
2.4.1 Language Settings ..................................................................................................... 20
2.4.2 Scheduled Reboot ..................................................................................................... 21
2.5 Tools and Information .......................................................................................................... 21
2.5.1 Reboot Tools .............................................................................................................. 21
2.5.2 Update Firmware ...................................................................................................... 21
2.5.3 Upload and Backup Configuration ............................................................................ 22
2.5.4 Restore Configuration ............................................................................................... 22
2.6 Information .......................................................................................................................... 23
3. MODULE ......................................................................................................................................... 24
3.1 MODULE Settings ................................................................................................................. 24
3.1.1 Call Duration Limit Settings ....................................................................................... 26
3.2 DTMF .................................................................................................................................... 29
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3.3 Toolkit ................................................................................................................................... 30
4. VOIP ................................................................................................................................................ 32
4.1 VOIP Endpoints ..................................................................................................................... 32
4.1.1 Add New SIP Endpoint .............................................................................................. 32
4.1.2 Add New IAX2 Endpoint ............................................................................................ 39
4.2 Batch SIP Endpoints .............................................................................................................. 45
4.3 Advanced SIP Settings .......................................................................................................... 46
4.3.1 Networking ................................................................................................................ 46
4.3.2 Paesing and Compatibility ......................................................................................... 50
4.3.3 Security ...................................................................................................................... 52
4.3.4 Media ........................................................................................................................ 53
4.3.5 Codec Settings ........................................................................................................... 54
4.4 Advanced IAX2 Settings ........................................................................................................ 55
4.4.1 General Settings ........................................................................................................ 55
4.4.2 Music on Hold ........................................................................................................... 56
4.4.3 Instruction of Codec Settings .................................................................................... 57
4.4.4 Jitter Buffer Settings .................................................................................................. 58
4.4.5 Misc Settings ............................................................................................................. 59
4.4.6 Quality of Service ...................................................................................................... 60
5. Routing ........................................................................................................................................... 60
5.1 Groups .................................................................................................................................. 64
5.2 Batch Creating rules ............................................................................................................. 65
5.3 MNP Settings ........................................................................................................................ 66
6. SMS ................................................................................................................................................. 67
6.1 General ................................................................................................................................. 67
6.1.1 Sender Options .......................................................................................................... 67
6.1.2 SMS to Email ............................................................................................................. 67
6.1.3 SMS Control ............................................................................................................... 69
6.1.4 HTTP to SMS .............................................................................................................. 71
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6.1.5 SMS to HTTP .............................................................................................................. 71
6.2 SMS Sender .......................................................................................................................... 71
6.3 SMS Inbox ............................................................................................................................. 72
6.4 SMS Outbox .......................................................................................................................... 72
6.5 SMS Forwarding ................................................................................................................... 73
7. Network .......................................................................................................................................... 74
7.1 LAN Settings ......................................................................................................................... 74
7.2 WAN Settings ........................................................................................................................ 76
7.3 VPN Settings ......................................................................................................................... 77
7.4 DDNS Settings....................................................................................................................... 78
7.5 Toolkit ................................................................................................................................... 78
7.5.1 Ping and Traceroute .................................................................................................. 78
7.5.2 TCP Capture ............................................................................................................... 79
7.6 Security Settings ................................................................................................................... 80
7.6.1 Firewall Settings ........................................................................................................ 80
7.6.2 White/Black List Settings ........................................................................................... 80
7.7 Security Rules ....................................................................................................................... 82
7.8 SIP Capture ........................................................................................................................... 83
8. Advances ........................................................................................................................................ 84
8.1 Asterisk API ........................................................................................................................... 84
8.2 Asterisk CLI ........................................................................................................................... 86
8.3 Asterisk File Editor ................................................................................................................ 87
8.3 Cloud Management .............................................................................................................. 87
9. Logs ................................................................................................................................................ 89
Appendix Feature List ......................................................................................................................... 91
General Info ................................................................................................................................ 91
VOIP Characters .......................................................................................................................... 92
Network ...................................................................................................................................... 92
System Features ......................................................................................................................... 93
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Management .............................................................................................................................. 94
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1. Overview
1.1 What is SWG-2016/32？
OpenVox SWG-2016/32 series wireless gateways include SWG-2016 G/C/L and SWG-2032 G/C/L,
which can provides 16/32 GSM/CDMA/WCDMA/LTE channels. They can bring you excellent HD
voice service with multiple codecs, including G.711U, G.711A, GSM, G.722, G.723, G.726, G.729,
and also flexible SMS service with multiple SMS API. The SWG-2016/32 series gateways will be
100% compatible with Asterisk, 3CX, FreePBX, FreeSWITCH and VOS VoIP operating platform, and
help users reduce telecommunications and communication costs.
1.2 Product Introduction
The SWG-2016/32 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
CDMA 2000: 800MHz
1 1 1
SWG-2016G
GSM
16
2
GSM: 850/900/1800/1900MHz
1 1 1
SWG-2016L
LTE
16
2
China/India
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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Europe/Middle East/Africa/
Korea/Thailand
LTE FDD: B1/B3/B5/B7/B8/B20
LTE TDD: B38/B40/B41
WCDMA: B1/B5/B8
GSM: B3/B8
SWG-2032C
CDMA
32
2
CDMA 2000: 800MHz
1 1 1
SWG-2032G
GSM
32
2
GSM: 850/900/1800/1900MHz
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
Europe/Middle East/Africa/
Korea/Thailand
LTE FDD: B1/B3/B5/B7/B8/B20
LTE TDD: B38/B40/B41
WCDMA: B1/B5/B8
GSM: B3/B8
1.3 Application
1.3.1 LCD And Buttons
LED Indicator/Icon/Buttons
Color/ Icon
Staus
Display Icon
Module Initiating,Disable
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No SIM Card
Searching for Signal
One grid Signal
Two grid Signal
Three grid Signal
four grid Signal
fives grid Signal
Worst Signal Quality During a Call
Medium Signal Quality During a Call
Best Signal Quality During a Call
Network Status LED
Green and
Flash
Network Connected
PWR
Always Green
Power on
POWER 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.
1.3.2 Multifunction button
1. [▲] : Press this key to flip up
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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.
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The main factions are as follows:
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1.3.3 Console
To ensure easy maintenance, SWG-2016/32 series gateway devices provide a serial port with a
baud rate of 115200 bps. Users can connect to the computer through RJ45 to USB cable for
maintenance related configuration.
Login device:
Step 1: Prepare the following serial cable (baud rate: 115200bps)
Step 2: Connect the USB port of the serial cable to the PC; connect the RJ45 port to the console
port of the device.
Step 3: Configure the login software
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After the above configuration, click "Open" to enter the device's background page. Use the same
login name and password as SSH to enter the system.
1.4 Main Features
Based on Asterisk®
Wide selection of codecs and signaling protocol
Support SMS sending, receiving, group sending
Support transferring SMS to E-mail
Support SMS remotely controlling gateway
Support USSD service
Support PIN identification
Support unlimited routing rules and flexible routing settings
SIM cards are all hot-swap
Stable performance, flexible dialing, friendly GUI
1.5 Physical Information
Size(No antenna and hanging ears): 440mm*44mm*300mm
LCD dimension:2.4"
LCD resolution ratio: 240*400
LAN port:1
WAN port:1
USB Interface:1
TF Infterface:1
SIM Cards: hot-swap
Operation Temperature: 0~40°C
Storage Temperature: -20~70°C
Operation humidity:10% ~ 90% non-condensing
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1.6 Software
Default IP:172.16.98.1
Username:admin
Passward:admin
For first time, you can access SWG-1016C using default IP 172.16.98.1. Then configure the module
as you want.
2. System
2.1 Status
On the “Status” page, you will find all Modules, SIP, IAX2, Routing and Network information.
Figure 2-1 Systm Status
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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
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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.
2.2 Time
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.
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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.
2.3 Login Settings
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.
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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.
2.4 General
2.4.1 Language Settings
You can choose different languages for your system. If you want to change 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:
Figure 2-4 Language Settings
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2.4.2 Scheduled Reboot
If 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 and By Running Time”.
Figure 2-5 Reboot Type
If use your system frequently, you can set this enable, it can helps system work more efficient.
2.5 Tools and Information
2.5.1 Reboot Tools
You can choose system reboot and asterisk reboot separately.
Figure 2-6 Reboot Tools
If you press "OK", your system will reboot and all current calls will be dropped. Asterisk Reboot
is the same.
2.5.2 Update Firmware
We offer 2 kinds of update types for you, you can choose System Update or System Online Update.
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If you choose System Online Update,you will see the following information:
figure 2-7 Update Firmware
2.5.3 Upload and Backup Configuration
If you want to update your system and remain your previous configuration, you can first
backup configuration, then you can upload configuration directly. That will be very convenient for
you.
Figure 2-8 Upload and Backup Configuration
2.5.4 Restore Configuration
Sometimes there is something wrong with your gateway that you don’t know how to solve
it, mostly you will select factory reset. Then you just need to press a button, your gateway will
be reset to the factory status.
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Figure 2-9 Restore Configuration
2.6 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 help information.
Figure 2-10 Information
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3. MODULE
3.1 MODULE Settings
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
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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.
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SMS Center Number
Your SMS center number of your local carrier.
Module IMEI
Only CDMA module does not support modifying IMEI
3.1.1 Call Duration Limit Settings
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,
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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
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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.
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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
3.2 DTMF
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
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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.
3.3 Toolkit
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.
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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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4. VOIP
4.1 VOIP Endpoints
This page shows everything about your SIP&IAX2, you can see 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 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.)
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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
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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
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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.
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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).
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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.
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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
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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.
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 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
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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.
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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
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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
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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
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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
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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).
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
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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.
4.3 Advanced SIP Settings
4.3.1 Networking
Networking General
Figure 4-16 Networking General
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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".
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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
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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
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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
4.3.2 Paesing and Compatibility
Figure 4-19 Paesing and Compatibility
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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
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attempts before we give up. 0 = continue
forever, hammering the other server until it accepts the
registration. Default is 0 tries, continue forever.
4.3.3 Security
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
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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.
4.3.4 Media
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
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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
4.3.5 Codec Settings
Select codecs from the list below.
Figure 4-22 Codec Settings
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4.4 Advanced IAX2 Settings
4.4.1 General Settings
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)
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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.
4.4.2 Music on Hold
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.
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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)
4.4.3 Instruction of Codec Settings
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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4.4.4 Jitter Buffer Settings
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.
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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.
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.
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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.
4.4.6 Quality of Service
Figure 4-28 Quality of Service
Table 4-22 Instruction of Quality of Service
Options
Definition
Tos
Type of service
Cos
Class of service
5. Routing
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
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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
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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
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[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 match
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 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
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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”.
5.1 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.
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Figure 5-5 Routing Group
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
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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.
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
6.1 General
You can choose enable SMS Received, SMS Local Strored and SMS Status Report or not.
Figure 6-1 SMS Settings
6.1.1 Sender Options
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.
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
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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.i
n
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.
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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.
6.1.3 SMS Control
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.
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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
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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
6.1.4 HTTP to SMS
Figure 6-5 HTTP to SMS
6.1.5 SMS to HTTP
Figure 6-6 SMS to HTTP Settings
6.2 SMS Sender
You can choose one or more ports to send SMS to the destination number, different numbers
should be separated by symbols: '\r', '\n', space character, semicolon and comma. Then you can
see much feedback information.
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Figure 6-7 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.
Figure 6-8 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.
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Figure 6-9 SMS Outbox
6.5 SMS Forwarding
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.
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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.
7. Network
7.1 LAN Settings
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.
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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.
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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.
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 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).
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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.
7.3 VPN Settings
SWG-2016/32 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
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Account
Server account
Password
The server's password
Use MPPE
Whether to use MPPE
Connection Status
Is it successful to connect to the server
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.
7.5 Toolkit
7.5.1 Ping and Traceroute
It is used to check network connectivity. Support Ping command on web GUI.
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Figure 7-5 Toolkit
7.5.2 TCP Capture
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
7.6.1 Firewall Settings
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.
7.6.2 White/Black List Settings
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
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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
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7.7 Security Rules
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
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If you see windows like below. It means your configuration has been applied successfully.
Figure 7-13 Security Rules Apply
7.8 SIP Capture
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
8.1 Asterisk API
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
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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.
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Figure 8-2 Telnet Access Gateway API
8.2 Asterisk CLI
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
On this page, you are allowed to edit and create configuration files. Click the file to edit.
Figure 8-4 Asterisk File Editor
Click “New Configuration File” to create a new configuration file. After editing or creating,
please reload Asterisk.
8.3 Cloud Management
SWG-2016/32 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.
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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
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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
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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.
Appendix Feature List
General Info
LAN:1
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WAN:1
Console:1
USB Interface:1
TF Interface:1
LCD dimension:2.4"
LCD resolution ratio: 240*400
SIM Cards: hot-swap
Temperature: -20~70°C (Storage) 0~40°C (Operation)
Operation humidity: 10% ~ 90%non-condensing
VOIP Characters
Support SIP, IAX2 Protocol
Add, Modify & Delete SIP/IAX2 Trunk
SIP/IAX2 Registration with Domain
Combine Different SIP/IAX2 Trunk into Group
DTMF Mode: RFC2833/Inband/SIPInfo
SIP V2.0 RFC3261 Compliance
Multiple SIP/IAX2 Registrations modes:
None (No registration, just IP and Password authenication)
Endpoint registers with this gateway (work as a SIP Sever)
This gateway registers with the endpoint (work as a SIP/IAX2 client)
Network
IPv4，UDP/TCP，DHCP，TELNET，HTTP/HTTPS，TFTP
PPTP VPN
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HTTP/SSH（Optical Telnet）
Ping & Traceroute Command on the Web
Simple Security Strategy: white list, black list, security rules
System Features
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
PIN Identification
Optional Voice Codec
Ports Group Management
SMS Bulk Transceiver, Sent to Email and Automatically Resend
SMS Coding/Detecting Automatically Identification
SMS Remotely Controlling Gateway
SMS Forwarding and Quick Reply
USSD transceiver
Outbound
Automatically Reboot
Support MMP
Support for custom scripts, dialplans
Support Openvox cloud manage
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Management
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
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