3G UMTS User Manual
Ope nVox Communication Co.Ltd URL:www.op envox.cn
3G Gateway User Manual
Address: 10/F, Building 6-A, Baoneng Science and Technology Industrial Park, Longhua New District,
Shenzhen, Guangdong,China 518109
Tel: +86-755-82535461, 82535095, 82535362
Fax: +86-755-83823074
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
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Version1.0 (2015-11-10)
Full text
The overall layout adjustment
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Copyright
Copyright© 2013 OpenVox Inc. All rights reserved. No part of this document may be reproduced without
prior written permission.
Confidentiality
Information contained herein is of a highly sensitive nature and is confidential and proprietary to OpenVox
Inc. No part may be distributed, reproduced or disclosed orally or in written form to any party other than
the direct recipients without the express written consent of OpenVox Inc.
Disclaimer
OpenVox Inc. reserves the right to modify the design, characteristics, and products at any time without
notification or obligation and shall not be held liable for any error or damage of any kind resulting from the
use of this document.
OpenVox has made every effort to ensure that the information contained in this document is accurate and
complete; however, the contents of this document are subject to revision without notice. Please contact
OpenVox to ensure you have the latest version of this document.
Trademarks
All other trademarks mentioned in this document are the property of their respective owners.
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Contents
1. Overview...................................................................................................................................... 1
What is 3G Gateway? 1
Sample Application 1
Product Appearance 2
Main Features 4
Physical Information 4
Software 5
2. System .............................................................................................................................................. 6
Status 6
Time 7
Login Settings 8
General and Cluster 10
Language Settings 10
Scheduled Reboot 10
Working Mode 11
Tools and Information 12
Reboot Tools 12
Information 14
3. MODULE......................................................................................................................................... 15
Module Settings 15
IMEI Modification 16
Call Duration Limit Settings 19
Call Forwarding 22
Call Waiting 23
DTMF 24
Toolkit 25
4. VOIP ............................................................................................................................................... 27
VOIP Endpoints 27
Advanced: Registration Options 29
Call Settings 30
Advanced: Signaling Settings 30
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Advanced: Timer Settings 31
Advanced SIP Settings 32
Networking 32
Advanced: NAT Settings 33
Parsing and Compatibility 35
Security 36
Media 37
Codec Settings 37
Advanced IAX2 Settings 38
5. Routing........................................................................................................................................... 41
Call Routing Rule 41
DISA Settings 42
Advance Routing Rule 42
Groups 45
MNP Settings 45
6. SMS ................................................................................................................................................ 47
SMS Settings 47
General 47
Sender Options 47
SMS to E-mail 48
SMS Control 50
HTTP to SMS 51
SMS to HTTP 51
SMS Sender 52
SMS Inbox 52
7. Network ......................................................................................................................................... 54
LAN Settings 54
DDNS Settings 56
Toolkit 56
8. Advanced ....................................................................................................................................... 58
Asterisk API 58
Asterisk CLI 60
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Asterisk File Editor 61
9. Logs ................................................................................................................................................ 62
Appendix Feature List........................................................................................................................ 67
Application Diagrams ........................................................................................................................ 70
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1. Overview
What is 3G Gateway?
OpenVox VoxStack Series 3G Gateway is an open source asterisk-based GSM VoIP Gateway
solution for SMBs and SOHOs. With friendly GUI and unique modular design, users may easily
setup their customized Gateway. Also secondary development can be completed through AMI
(Asterisk Management Interface).
There are three models with VoxStack series of 3G Gateway, the VS-GW1202, VS-GW1600 and VS-
GW2120. The Modular Design of 3G Gateways are ranging from 4 up to 44 3G/UMTS channels, it is
designed for multiple frequency bands Including 3G/UMTS 850/900/1900/2100 MHz, and it can
support a wide selection of codecs, including G.711A, G.711U, G.729, G.722, G.726. For a reliable
connecting 3G to voip gateway, it can quickly reduce telecommunication expenses and maximize
cost-savings. With the unique design of the VoxStack Gateway, it can support hot-swap for both
SIM cards and 3G gateway modules. Users can simply add or remove the modules for hardware
expansion or exchange. Each 3G gateway module runs an independent asterisk system inside. The
VoxStack gateway designs with two LAN Switch boards that provide stack ability on the hardware
upgrade. The 3G Gateways will be 100% compatible with asterisk, Elastix, trixbox, 3CX ,
FreeSWITCH SIP server and VOS VoIP operating platform.
Sample Application
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Figure 1-2-1TopologicalGraph
Product Appearance
The picture below is appearance of VS-GW2120.
Figure 1-3-1 Product Appearance
Figure 1-3-2 Front Panel
Reset Reset
Reset
Reset
Reset
0
4
1
2
5
3
ETH1
ETH2
○
0 Network Data Switch Board: ETH1, ETH2.
ETH1: Access Board 1 only, provide proxy access to other boards with different port numbers
which can avoid IP conflict.
ETH2: Access different Boards with different IP address.
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VoxStack provides 2 working modes: Stand-alone and Cluster.
Stand-alone: A single IP address manages one 3G modules (4 ports).
Table1-3-1ETH2 IP Addresses
Stack Num
IP
Username
Password
○
1
172.16.99.1
admin
admin
○
2
172.16.99.2
admin
admin
○
3
172.16.99.3
admin
admin
○
4
172.16.99.4
admin
admin
○
5
172.16.99.5
admin
admin
Cluster: A single IP address manages up to 5 GSM modules (up to 20 ports).
Default IP: 172.16.99.1
Figure 1-3-3 Front Panel
Reset Reset
Reset
Reset
ETH1
ETH2
Reset
Table 1-3-2 LED Indicator
LED Indicator
Color
Status
⑥Signal Status LED
Green and Flash
Module Initiating
Red and Flash
No SIM Card
Red and No-flash
Worst Signal Quality
Yellow and No-flash
Medium Signal Quality
Green and No-flash
Best Signal Quality
⑦Call Status LED
Flash (0.25s)
Communicating
Blind
Normal
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⑧⑨Network Status LED
Green and Flash
Network Connected
⑩Running Status LED
Green and Flash(1s)
Work Normally
⑾Power LED
Always Green
Supply Power
Main Features
Modular and VoxStack design
Based on Asterisk
○R
Editable Asterisk
○R
configuration file
Wide selection of codecs and signaling protocol
Support SMS sending, receiving, group sending
Support transferring SMS to E-mail
Support SMS automatically resent
Support SMS remotely controlling gateway
Support USSD service
Support IMEI modification
Support PIN identification
Support unlimited routing rules and flexible routing settings
SIM cards and modules are all hot-swap
Stable performance, flexible dialing, friendly GUI
Physical Information
Table 1-5-1Description of Physical Information
Weight
6144g(VS-GW2120-28w)
Size
44cm*34cm*9cm
Frequency
GSM 850/900/1800/1900MHz
Temperature
-40~85°C (Storage)
-40~85°C (Operation)
Operation humidity
10%~90% non-condensing
Max power
95W
LAN port
2
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Software
Default IP: 172.16.99.x
Username: admin
Password: admin
Every VS-GWM400G is independent with each other, please enter the default IP in your browser to
scan and configure the module you want. For the first time, please enter every IP to configure
every module one by one. Now we offer you two ways to access to your gateway. There are two
RJ45 Network ports on the board, ETH1 and ETH2. They are different. If chooseETH1, you can
access Board 1 only, then access to other boards with different port numbers but the same IP
address, this will help to avoid IP conflict. If you choose ETH2, you can access different Boards with
different IP addresses .
Notice: Log in
Figure 1-6-1 LOG IN Interface
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2. System
Status
On the “Status” page, you will find all GSM, SIP, Routing, Network information and status.
Figure 2-1-1 System Status
Table 2-1-1 Description of System Status
Options
Definition
Port
Number of GSM ports.
Signal
Display the signal strength of in each channels of GSM.
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BER
Bit Error Rate.
Carrier
Display the network carrier of current SIM card.
Registration Status
Indicates the registration status of current GSM 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
networkis required to play an announcement before completing the call
then thisdefinition of PDD excludes the duration of such announcements.
ACD
The Average Call Duration (ACD) is calculated by taking the sum ofbillable
seconds (bill sec) of answered calls and dividing it by the numberof 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.
Module Status
Show the status of port, include blank space and “READY”. Black space
means it is unavailable here and “Ready” means the port is available.
Remain Time
This value is multiplied by to step length is a rest call time.
Time
Table 2-2-1Description 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 timezone 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].
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NTP Server 3
The second reserved NTP server. For example, [time.nist.gov].
Auto-Sync from NTP
Whether enable automatically synchronize from NTP server or not. ON
（enable）, OFF （disable）.
Sync from NTP
Sync time from NTP server.
Sync from Client
Sync time from local machine.
For example, you can configure like this:
Figure 2-2-1 Time Settings
You can set your gateway time Sync from NTP or Sync from Client by pressing different buttons.
Login Settings
Your gateway doesn't have administration role. All you can do here is to reset what new username
and password to manage your gateway. And it has all privileges to operate your gateway. You can
modify “Web Login Settings” and “SSH Login Settings”. If you have changed these settings, you
don’t need to log out, just rewriting your new user name and password will be OK. Also you can
specify the web server port number.
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Table 2-3-1Description of Login Settings
Options
Definition
User Name
NOTES: Your gateway doesn’t have administration role.
All you can do here is defining the username and password to manage
your gateway.
And it has all privileges to operate your gateway.
User Name: 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.
Port
Specify the web server port number.
Figure 2-3-1 Login Settings
Notice: Whenever you do some changes, do not forget to save your configuration.
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General and Cluster
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”.
Figure 2-4-1 Language Settings
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 2-4-2 Reboot Types
If use your system frequently, you can set this enable, it can helps system work more efficient.
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Working Mode
OpenVox 3G Gateway offers you two ways to cluster your gateway: Automatic Cluster or Manual
Cluster.
When you first time log in your gateway, you will only see 4 ports of one module. Then you can
press button, the system will search other modules in the LAN and
communicate.
Figure 2-4-3 Automatic Cluster
If you want to choose Manual Cluster, you should switch Detail on first.
There are five VS-GW2120 modules which are independent with each other integrated in one 3G
Gateway, so we offer 3 kinds of Working Modes.
Figure 2-4-4 Working Modes
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Stand-alone Mode: Run alone, total 4 ports.
Master Mode: Run as master with two different IP, controlling up to 10 slaves.
(The master can be accessed by the original IP. The target IP is used to communicate with the
slaves.)
Slave Mode: Run as slave with two different IP, controlled by the master. If the original IP is
forbidden, the slave can be accessed by the master with inward IP only.
Notice: You can choose Remain Original IP address ON or OFF. If set it on, you can log in your getaway
with Original IP and Target IP.
Table 2-4-1 Definition of Master Options
Options
Definition
Mode
Stand-alone Mode:
Run alone total 4 ports
Master Mode:
Run as master with two different IP, controlling up to 10 slaves.
The master can be accessed by the original IP. The target IP is used to
communicate with the slaves.
Slave Mode:
Run as slave with two different IP,controlled by the master.
If the original IP is forbidden, the slave can be accessed by the
master with inward IP only.
Password
(master mode)
Master Mode password.
Must be 4~16bits digital 0-9.
Master IP(Local IP)
(master mode)
Master’s target IP.
Must be set in the subnet different from external subnet,
so that the external subnet couldn’t access internal subnet.
Slaves IP List
Set the slaves’s original and target IP. The original IP is outward IP, and the
target IP is inward IP. Up to four slaves.
Tools and Information
On the “Tools” pages, there are reboot, update, upload, downloadand reset toolkits.
Reboot Tools
You can choose system reboot and Asterisk reboot separately.
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Figure 2-5-1 Reboot Prompt
If you press “Yes”, your system will reboot and all current calls will be dropped. Asterisk Reboot is
the same.
Table 2-5-1 Instruction of reboots
Options
Definition
System Reboot
This will turn off your gateway and then turn it back on. This
will drop all current calls.
Asterisk Reboot
This will restart Asterisk and drop all current calls.
We offer 2 kinds of update types for you, you can choose System Update or System Online Update.
System Online Update is an easier way to update your system, if you choose it, you will see some
information below.
Figure 2-5-2Prompt Information
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.
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Figure 2-5-3 Upload and Download
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.
Figure 2-5-4 Factory Reset
Information
On the “Information” page, there shows some basic information about the 3G gateway. You can
see software and hardware version, storage usage, memory usage and some help information.
Figure 2-5-5 System Information
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3. MODULE
You can see much information about your SIM cards on this page.
Module Settings
Figure 3-1-1 Module Settings
On this page, you can see your 3G module status and click action button to configure the
port.
Figure 3-1-2 Port Configure
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As you see, we have offered “Band” option, you can select different bands easily and you have
many options.
If you have set your Pin Code, you can check on like this:
Figure 3-1-4 PIN Code Application
Then input your password, system will identify numbers of SIM cards. It can help to prevent SIM
card from being stolen and improve security.
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).
Figure3-1-5 CLIR Application
When you are on the phone, other calls coming in, you can set Ca ll Waiting on, then the coming
calls won’t be hung up.
IMEI Modification
One more feature, we offer you IMEI automatically modification.
Figure 3-1-8 Automatically IMEI Modify
We have offered you IMEI modification function. If you want to modify your IMEI number, please
do as follows.
You can log in your gateway and modify IP address as follows. Input web site below on your
browser. http://your gatewaIp/cgi-bin/php/gsm-autoimei.php.
Then you will see the following picture. Don’t forget to switch “Enable” to “ON”, or you can’t
change your IMEI numbers.
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Figure 3-1-9 IMEI Modification
Also you can choose to modify one or more certain ports or all ports. You can set automatic
modification interval by filling in the time you want.
If you choose “Immediately”, then the ports you have chosen will modify IMEI numbers at once.
On the contray, system will keep time from now until the time of next modification. And If you
choose “Force”, System will hang up all your current calls, then modify IMEI.
You can press to do some settings. We offer you two ways to
modify your IMEI. You can choose Autogeneration or Manual.
Figure 3-1-10 Advanced Settings
As you can see, you can set any number you wan for every port. “X” means any digits from 0 to 9.
You just need to fill in “Set to All ”, then press “Set to All”, you can see the interface as above. Don’t
forget to press “Save”. Then “Current IMEI” will change. That means Autogeneration. If you want to
set a certain number as your IMEI, you can press “Manual”. Then you will be required to input a
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new IMEI.
Figure 3-1-11 Manual
After configuration, you can press “Back Home” to return your gateway interface.
Table 3-1-1 Definition of GSM Settings
Options
Definition
Name
The alias of the GSM port. 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.
DAC Gain
The range is: -42 to +20
ADC Gain
The range is: -42 to +20
Dial Prefix
The prefix number of outgoing calls from this GSM 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.
GSM Module
IMEI
You can click “Modify” button and automatically modify it.
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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.
Figure3-1-12 Single Settings
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.
Call Duration Limitation
This will limit your total calling time of this port.
Figure 3-1-13 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”.
SMS Center
Number
Your SMS center number of your local carrier.
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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 3-1-14 Auto Reset Settings
You can save your configuration to other ports.
Figure 3-1-15Save 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.
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Figure 3-1-16GSM Information
Table 3-1-2 Description of Call Duration Limit Settings
Options
Definition
Step
Step length value range is 1-120 s, 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 GSM channel. The
max call duration is between 1 to 999999 minutes.
Call Duration
Limitation
The value of limitation single call, this value range is 1-65535. Step
length multiplied by time of single call just said a single call duration
time allowed.
Minimum Charging
Time
A single call over this time, GSM 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.
Remain Time
This value is multiplied by to step length is a rest call time.
Enable Auto Reset
Automatic restore remaining talk time, that is, get total call minutes
of GSM channel.
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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
Call Forwarding
Sometimes it’s not convenient for you to answer a call, if you don’t want to lose some important
calls, you can choose Call Forwarding. You can choose Call Forwarding Unconditional, Call
Forwarding No Reply, Call Forwarding Busy or Call Forwarding on Not Reachable.
If want to cancel your call forwarding settings, you can choose Cancel All.
Figure 3-2-1Call Forwarding
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Notice: Don’t forget to save your settings. Please first press button, then press
button..
Call Waiting
Call Waiting is a feature that alerts you to an incoming call while you're on another call. It also lets
you switch between the calls. It's automatically included with all Verizon Wireless plans.
Figure 3-2-1Call Forwarding
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DTMF
You can do some DTMF Detection Settings if you choose “GSM -> DTMF”.
Figure 3-3-1 DTMF Detection Settings
Notice: If you don’t have special need, you don’t have to modify these settings. You can just choose
“Default”.
DTMF Normal Twist and Reverse Twist 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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Toolkit
You can get USSD information, send AT command and check number with this module. When you
have a debug of the GSM module, AT command is useful.
Figure 3-4-1 Function Options
Table 3-4-1 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.
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AT Command
To perform some specific AT commands. This is useful when you have a
debug of the GSM 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 3-4-2 AT Command Example
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4. VOIP
VOIP Endpoints
This page shows everything about your SIP, you can see status of each SIP.
Figure 4-1-1 SIP Endpoints
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. You can choose Anonymous, Endpoint registers
with this gateway or This gateway registers with the endpoint.
You can configure as follows:
If you set up a SIP endpoint by registration “None” to a server, then you can’t register other SIP
endpoints to this server. (If you add other SIP endpoints, this will cause Out-band Routes and
Trunks confused.)
Figure 4-1-2 Anonymous Registration
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For convenience, we have designed a method that you can register your SIP endpoint to your
gateway, thus your gateway just work as a server.
Figure 4-1-3 Register to Gateway
Also you can choose registration by “ This gateway registers with the endpoint”, it’s the same with
“None”, except name and password.
Figure 4-1-4 Register to Server
Table 4-1-1 Definition of SIP Options
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Options
Definition
Name
A name which is able to read by human.
Username
User Name the endpoint will use to authenticate with the gateway.
Password
Password the endpoint will use to authenticating with the gateway.
Allow characters.
Registration
Whether this endpoint will register to this gateway or this gateway to the
endpoint.
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.
Notice: If the input here is hostname and your DNS has changed, you must
reboot asterisk.
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
Addresses NAT-related issues in incoming SIP or media sessing.
Advanced: Registration Options
Table 4-1-2 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.
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Port
The port number the gateway will connect to at this endpoint.
Quality
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 signaling instead of
sending signaling directly to endpoint.
Call Settings
Table 4-1-3 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.
Advanced: Signaling Settings
Table 4-1-4 Definition of Signaling Options
Options
Definition
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Progress Inband
If we should generate in-band ringing.
Always use ‘never’ to never use in-band signaling, even in cases where
some buggy devices might not render it.
Valid values: yes, no, never.
Default: never
Allow Overlap
Dialing
Whether or not the Remote-Party-ID header should be trusted.
Append
user=phone to URI
Whether or not to add ‘;user=phone’ to UPIs that contain a valid phone
number.
Add Q.850 Reason
Headers
Whether or not to add Reason header and to use it if is available.
Honor SDP Version
By default, the gateway will honor the session version number in SDP
packets and will only modify the SDP session if the version number
changes. Turn this option off to force the gateway to ignore the SDP
session version number and treat all SDP data as new data.
This is required for devices that send non-standard SDP packets (observed
with Microsoft OCS).
By default this option is on.
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 inca pable 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.
Advanced: Timer Settings
Table 4-1-5 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
Refresh Interval
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.
Advanced SIP Settings
Networking
Table 4-2-1 Definition of Networking Options
Options
Definition
UDP Bind Port
Choose a port on which to listen for UDP traffic.
Enable TCP
Enable server for incoming TCP connection (default is no).
TCP Bind Port
Choose a port on which to listen for TCP traffic.
TCP 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).
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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.
Advanced: NAT Settings
Table 4-2-2 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.
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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 Address:
The external address (and optional TCP port) of the NAT.
External Address=hostname[:port] specifies a static address [:port] to be
used in SIP and messages. Examples:
External Address=12.34.56.78
External Address=12.34.56.78:9900
External
Hostname
The external hostname (and optional TCP PORT) of the NAT.
External Hostname=hostname[:port] is similar to “External Address”.
Examples:
External Hostname=foo.dyndns.net
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.
Table 4-2-2 Definition of RTP S ettings
Start of RTP Port
Range
Start of range of port numbers to be used for RTP.
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End of RTP port
Range
End of range of port numbers to be used for RTP.
Parsing and Compatibility
Table 4-2-3 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, (o=).
This filed MUST NOT contain spaces.
Disallowed SIP
Methods
When a dialog is started with another SIP endpoint, the other endpoint
should include an Allow header telling us what SIP methods the endpoint
implements. However, some endpoints either do not include an Allow
header or lie about what methods they implement. In the former case,
the gateway makes the assumption that the endpoint supports all known
SIP methods. If you know that your SIP endpoint does not provide
support for a specific method, then you may provide a list of methods
that your endpoint does not implement in the disallowed_methods
option. Note that if your endpoint is truthful with its Allow header, then
there is no need to set this option.
Hangup Cause Code
Hangup Cause Code help
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).
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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
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.
Security
Table 4-2-4 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.
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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.
Media
Table 4-2-5 Instruction of Media
Options
Definition
TOS for SIP
Packets
Sets type of service for SIP packets
TOS for RTP
Packets
Sets type of service for RTP packets
Codec Settings
Select codecs from the list below.
Figure 4-2-1 Codec Settings
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Advanced IAX2 Settings
Table 4-3-1 Instruction of General Settings
Options
Definition
Bind Port
Bindport and bindaddr may be specified. (Default:4569)
Bind Address
More than once to bind to multiple addresses, but the first will be the default.
Enable
IAXCompat
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 IAX2
DPREQ (Dialplan Request) instead of blocking while it for a response.
Enable
Nochecksums
Disable UDP checksums (if nochecksums is set, then no checksums will be
calculated/checked on systems supporting this feature).
Enable Delay
Reject
You may specify a global default AMA flag for iaxtel calls. It must be one of
‘default’,’omit’,’billing’ or ‘documentation’. These flags are used in the generation of call
detail records.
ADSI
ADIS (Analog Display Service Interface) can be enabled if you have (or may have) ADSI
compatible CPE equipment.
SRV Loopup
Whether ro not to perform an SRV lookup on outbound calls.
AMA Flags
You ay specify a global default AMA flag for iaxtel calls. These flags are used in the
generation of call detail records.
Auto Kill
If we don’t get ACK to our NEW within 2000ms, and autokill is set to yes, then we cancel
the whole thing(that is 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, but would be ill advised 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 to specifying
on a per-user basis.
Table 4-3-2Instruction of Music On Hold
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Options
Definition
Mohsuggest
The ‘mohsuggest’ option specifies which music on hold class to suggest to the peer channel
when this channel places the peer on hold. It may be specified globally or on a per-useror
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 fallback to English（en）.
Table 4-3-3Instruction of Codec Settings
Options
Definition
Band Width
Specify bandwidth of low, medium, or high to control which codecs are used in general.
Disallow
Fine tune codecs here using “allow” and “disallow” clauses with specific codecs.
Allow
Fine tune codecs here using “allow” and “disallow” clauses with specific codecs.
Codec Priority
Codecpriority controls the codec negotiation of an inbound IAX2 call. This option is
inherited to all user entities. It can also be defined in each user entity separately which will
override the setting in general.
Table 4-3-4Instruction of Jitter Buffer Settings
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 do jitter buffering on
the switch, since the endpoints can each handle this. However, some endpoint may have
poor jitter buffers themselves, so this option will force *to always jitterbuffer, even in this
case.
Max Jitter
Buffers
A maximum size for the jitter buffer.
Resyncthreshold
When the jitterbuffer notices a significant change in delay that continues over a few frames,
it will resync, assuming that the change in delay was caused by a change in delay is
measured as twice the measured jitter plus resync threshold.
Max Jitter
Interps
The maximum number of interpolation frames the jitterbuffer should return in a row. Since
some clients do not send CNG/DTX frames to indicate silence, the jitterbuffer will assume
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silence has begun after returning this many interpolations. This prevents interpolation
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 plus 40
milliseconds. Increasing this value may help if your network normally has low jitter, but
occasionally has spikes.
Table 4-3-5 Instruction of Misc Settings
Options
Definition
IAX2 Thread
Count
Establishes the number of iax helper threads to handle I/O.
IAX2 Max
Thread Count
Establishes the number of extra dynamic threads that may be 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 dynamic peer after it completes registration.
MaxCallNumber
s_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 disable. Unlike the
‘maxcallnumbers’ option, this limit is not to 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 token user guide. This option’s default value of 8192
should be sufficient in most cases.
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5. Routing
Figure 5-1-1 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. shows current
routing rules. Otherwise you can set up unlimited routing rules.
Call Routing Rule
You can click button to set up your routings.
Figure 5-1-2 Example of set up Routing Rule
The figure above realizes that calls from “support” SIP endpoint switch you have registered will be
transferred to umts-1.1. When “Call Comes in From” is gsm, “prepend”, “prefix” and “match
pattern” in “Advanced Routing Rule” are ineffective, and just “CallerID” option is available.
Table 5-1-1 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, 'SIP2GSM' or 'GSM2SIP').
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Call Comes in
From
The launching point of incoming calls.
Send Call Through
The destination to receive the incoming calls.
DISA Settings
When you make “Authentication” switch to “ON”, this page is available.
Figure 5-1-3 DISA Settings
Table 5-1-2 Description of DISA Settings
Options
Definition
Authentication
enable or disable password authentication
Secondary Dialing
enable or disable secondary dialing
DISA Timeout
Select the timeout range of 1 to 10 seconds
Max Password
Digits
Restrict the max length of password
Password
Click the button to edit Authenticated Password
Advance Routing Rule
Figure 5-1-3 Advance Routing Rule
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Table 5-1-3 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 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 num ber matches.
Time Patterns that
will use this Route
Time Patterns that will use this Route help.
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Set the Caller ID
Name to
What caller ID name would you like to set before sending this call to the
endpoint .
Set the Caller ID
Number to
What caller number 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.
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 5-1-3Time 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 5-1-4Time Reference
You set your caller ID name and caller number as you like before sending the call to the endpoint.
You can also configure forward number when you have a transfer call.
Figure 5-1-5Change Rules
You can add one or more “Failover Call Through Numbers”.
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Figure 5-1-6 Failover Call Through Number
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 UMTS or SIP to groups. Then if you want to make a
call, it will find available port automatically.
Figure 5-2-1 Establish Group
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 ${num}.
Here is an example of the MNP url:
https://s1.bichara.com.br:8181/chkporta.php?user=832700&pwd=sdsfdg&tn=8388166902
The 8388166902 is the outgoing phone number, when config the MNP url, should replce it with
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${num}. Then it turns to
https://s1.bichara.com.br:8181/chkporta.php?user=832700&pwd=sdsfdg&tn=${num }.
Figure 5-3-1 MNP Settings
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6. SMS
SMS Settings
General
You can choose enable SMS stored and SMS status report or not. But if you want to see your SMS
outbox, you should switch SMS Status Report ON.
Figure 6-1-1 SMS Settings
Sender Options
You can change sender options here, include resend, times of resend.
Figure 6-1-2 Sender Options
Table 6-1-1 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.
Verbose
Verbose level of sending message
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SMS to E-mail
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 andsupport@openvox.cn
Figure 6-1-3 SMS to E-mail Options
Table 6-1-2 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
×
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Other: 163 free
e-mail
smtp.163.com
25
×
Table 6-1-3 Definition of SMS to E-mail
Options
Definition
Enable
ON(enable),OFF(disable)
SMTP Server
SMTP Server help
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.
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.
Title
Available variable:
$PHONENUMBR:SMS sender number
$PORT: SMS from which port.
$TIME: SMS received time.
$MESSAGE: SMS content.
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Content
Available variable:
$PHONENUMBR:SMS sender number
$PORT: SMS from which port.
$TIME: SMS received time.
$MESSAGE: SMS content.
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.
Figure 6-1-4 SMS Control
For example, SMS control password is 123456789 which has nothing to do with the login
password, you can send “get info 123456789” to the UMTS module’s phone number to get your
gateway’s IP information.
Table 6-1-4 Definition of SMS Control
Options
Definition
Enable
ON(enable), OFF(disable)
Password
The password to confirm that SMS makes the gateway rebooted, shut
down, restored configuration files and get info on this gateway.
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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
HTTP to SMS
Figure 6-1-5 HTTP to SMS
SMS to HTTP
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Figure 6-1-6 SMS to HTTP
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.
Figure 6-2-1 SMS Sender Status
SMS Inbox
On this page, you are allowed to scan, delete, clean up, and export each port’s received SMS. A lso
you are allowed to check messages by port, phone number, time order and message keywords.
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Figure 6-3-1 SMS Information
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7. Network
On “Network” page, there are five sub-pages, “LAN Settings”, “DDNS Settings”, and “Toolkit”.
LAN Settings
There are three types of LAN port IP, Factory, Static and DHCP. Factory is the default type, and it is
172.16.99.1. 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 7-1-1LAN Settings Interface
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Table7-1-1Definition 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.
Network
The subnet mask of your gateway.
Default Gateway
Default getaway IP address.
Reserved Access IP
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.
Enable
A switch to enable the reserved IP address or not.
ON(enabled), OFF(disabled)
Reserved Address
The reserved IP address for this gateway.
Reserved Netmask
The subnet mask of the reserved IP address.
Basically this info is from your local network service provider, and you can fill in four DNS servers.
Figure 7-1-2 DNS Interface
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Table7-1-2Definition of DNS Settings
Options
Definition
DNS Servers
A list of DNS IP address. Basically this info is from your local
network service provider.
DDNS Settings
You can enable or disable DDNS (dynamic domain name server).
Figure 7-2-1 DDNS Interface
Table7-2-1Definition 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.
Toolkit
It is used to check network connectivity. Support Ping command on web GUI.
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Figure 7-3-1 Network Connectivity Checking
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8. Advanced
Asterisk API
When you make “Enable” switch to “ON”, this page is available.
Figure 8-1-1 API Interface
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Table8-1-1Definition 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, 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. 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 172.16.123.123/255.255.0.0 is allowed to access the
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gateway API. Please refer to the following figure to access the gateway API by putty.
172.16.123.123 is the gateway’s IP, and 5038 is its API port.
Figure 8-1-2 Putty Access
Asterisk CLI
In this page, you are allowed to run Asterisk commands.
Figure 8-2-1 Asterisk Command Interface
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Table8-2-1Definition of Asterisk API
Options
Definition
Command
Type your Asterisk CLI commands here to check or debug your
gateway. e.g, type ”help” or ”?” you will get all help information.
If you type “help” or “?” and execute it, the page will show you the executable commands.
Asterisk File Editor
On this page, you are allowed to edit and create configuration files.
Click the file to edit.
Figure 8-3-1 Configuration Files List
Click “New Configuration File” to create a new configuration file. After editing or creating, please
reload Asterisk.
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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 9-1-1System Logs Control
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Figure 9-1-2 System Logs Output
You can choose one specific board to see it related logs.
Figure 9-1-3 Choose One Board
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Figure 9-1-4 AT Commands
Figure 9-1-5 CDR Output
Recently we have made our LOGS display richer, you can see your GSM Outbound of every port
clearly.
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Figure 9-1-5Time Patterns that will use this Route
Table9-1-1Definition 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.<br>
switch off :
logs will remain, and the file size will increase gradually.
default on, default size =1MB.
Verbose:
Asterisk console verbose message switch.
Notice:
Asterisk console notice message switch.
Warning:
Asterisk console warning message switch.
Debug:
Asterisk console debug message switch.
Error:
Asterisk console error message switch.
DTMF:
Asterisk console DTMF info switch.
Auto clean:
(asterisk 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, default size =100KB.
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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. <br>
default on, default size =100KB.
IAX Logs
Whether enable or disable IAX logs.
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, default size=100KB.
AT Commands Logs:
Displaying GSM module AT messages.
Auto clean:
(AT 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.<br>
switch off :
logs will remain, and the file size will increase gradually.
default on, default size =100KB.
Call Detail Record:
Displaying Call Detail Records for each channel.
Append IMEI
Switch on: IMEI will be appended to the CDR gsm channel in ‘From’
or ‘To’.
Switch off: No appended IMEI.
Default off.
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.<br>
switch off :
logs will remain, and the file size will increase gradually.
default on, default size =20MB.
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Appendix Feature List
Wireless port：GSM 850/900/1800/1900MHz
Storage temperature：-20~70℃
Operating temperature：0~40℃
Operating humidity：10％~90％
Power supply specifications：100-240V/1-2A
Maximum power：46W
Support Volume adjustment
Support Gain adjustment
Support CLID display & hide (need operators support)
Support IMEI number automatically modify, avoid blocking
Support PIN identification
Support GSM frequency range：850/900/1800/1900MHz
Support Band binding
Support SMS bulk transceiver, messages sent to email and SMS automatically resent
Support USSD transceiver
Support call transfer
Support open API interface（AMI）
Support add, modify & delete SIP trunk
Support different SIP trunks combined into one SIP trunk Group
Support SIP registration with domain
Support multiple SIP registrations：Anonymous, Endpoint registers with this gateway, This
gateway registers with the endpoint
Support protocols：SIP, IAX（easily expand other protocols that Asterisk supports）, TCP,
UDP, RTP, TELNET, HTTP, SMTP, POP3
Support routing before number changed
Support routing after number changed
Support GSM—>IP routing
General Info
GSM
SIP Features
Routingand N umber Conversion
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Support IP—>GSM routing
Support add, modify & delete routing
Support unlimited routing rules
Flexible routing settings
Permit and prohibit certain calls
Extensible automatic callback
Extensible speed dial
Support customizable IVR, DISA and other applications
Simple and convenient configuration via Web GUI
Signal strength indicator, LED status indicator
Support NTP time synchronization and client time synchronization
Support DTMF：RFC2833, In-band, Info
Working mode: stand-alone, Master, slave.
Each Board integrated independent Asterisk system would make it easier for independent
operation, improve stability
Support SSH access for background management, Asterisk CLI command operation
Support AT command interface
GSM ports group management
Support PDD (Post DialingDelay) display, ACD (Average Call Duration) display, ASR
(Answer-Seizure Ratio) display and BER (Bit Error Rate) display
Support configuration file backup and upload
Support for custom scripts, dialplans
Support codecs：G.711A, G.711U, G.729, G.722, G.723, G.726, GSM
Support DDNS
2*RJ45 Network port
① ETH1：access Board1 only, provide proxy access to other boards with different
port numbers which can avoid IP conflict
② ETH2: access different Boards with different IP addresses
Network type：Static IP, Dynamic, PPPoE
Support modify username and password for web login
Business Control
System Features
Network Features
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Support ping &traceroutecommand on the web
Support SMS remotely controllinggateway
SupportEcho cancellation, Jitter buffer
Support gateway boards and SIM hot plug
Reboot settings（According to the running time, specific time）
Update Firmware：HTTP
Support multiple detailed LOG output
Support Restore the factory settings
One YearWarranty
Operation Maintenance
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Application Diagrams
Thank You for Choosing OpenVox Products!
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