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.
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.
This document applies to all Mag1000 series analog gateways, including FXS/FXO. Different types of gateways may have functional differences. For details, please contact OpenVox sales or technical support.
All other trademarks mentioned in this document are the property of their respective owners.
OpenVox MAG 1000 is a multifunctional analog gateway for seamless connection between IPPBX, fax machines, analog phones and operators. It provides three interfaces of RJ11/RJ45/RJ21, covering almost all the wiring needs of users in all scenarios. In addition, it has excellent full concurrent voice/ fax processing capabilities, strong performance and high stability, and provides high-quality call services for operators, enterprises, call centers and residential users in residential communities.
The MAG series analog gateway consists of six models: support 16 FXS ports, support 24 FXS ports, support 32 FXS ports, support 16 FXO ports, support 24 FXO ports, support 32 FXO ports; support at least 16 ports, Maximum 32 ports, multiples of 8, FXO/FXS combination.
The MAG1000 Analog Gateways are developed for interconnecting a wide selection of codecs including G.711A, G.711U, G.729A, G.722, G.726, iLBC. MAG1000 series use standard SIP protocol and compatible with Leading VoIP platform, IPPBX and SIP servers, such as Asterisk, Issabel, 3CX, FreeSWITCH ,BroadSoft and VOS VoIP operating platform.
1.2 Sample Application
Figure 1-2-1 shows a simple application of the analog gateway series.
Figure 1-2-1 Topological Graph
1.3 Product Appearance
The picture below is appearance of MAG1000 Series Analog Gateway.
Figure 1-3-1 Product Appearance
Figure 1-3-2 Front Panel
1：Running status indicator
2：Power status indicator
3：Channel status indicator
Figure 1-3-3 Back Panel
9：RJ21 1-24 port
10：RJ21 25-32 port
1.4 Main Features
Support NTP time synchronization and client time synchronization
Support to modify username and password of web login
Whether enable automatically synchronize from NTP server or not. ON is enable, OFF is disable this function.
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.
2.3 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 both your “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.
Table 2-3-1 Description of Login Settings
Define your username and password to manage your gateway, without space here. Allowed characters
Please input the same password as 'Password' above.
Select the mode of login.
Specify the web server port number.
Specify the web server port number.
SSH login port number.
Figure 2-3-1 Login Settings
Notice: Whenever you do some changes, do not forget to save your configuration.
2.4.1 Language Settings
On our gateway products, you can set different languages according to your needs.
First, you need to turn on the “Advanced” mode.
And then “Download” the current language pack of the system.
Then click the "Browse" option.
After importing the language pack you need, click the "Add" button.
And it will take effect without restarting the gateway.
Figure 2-4-1 Language Settings
2.4.2 Scheduled Reboot
You can enable the automatic restart function to make your gateway restart after working for a certain period of time to achieve higher work efficiency.
Figure 2-4-2 Scheduled Reboot
In the tool page, users can restart the gateway, upgrade firmware, upload and back up configuration files, and restore factory settings.
The analog gateway supports individual system restart or Asterisk restart.
Figure 2-5-1 Reboot Tools
Notice: When you confirm the restart, the system will automatically end all current calls.
Table 2-5-1 Description of Reboots
This option will restart your gateway and cut off all current sessions.
This option will restart Asterisk and cut off all current sessions.
The analog gateway provides two firmware upgrade methods, you can choose system upgrade or system online upgrade. To select the system upgrade, you need to download the relevant firmware from the OpenVox website first. The system online upgrade is an eaiser way with one-click upgrade.
Figure 2-5-2 Update Firmware
After configuring your gateway, you can download the current configuration file. When you need to configure other gateways of the same model or restore the gateway to factory settings, you can choose to upload this backup configuration file without the need to reconfigure the gateway .
Notice：It will take effect only if the version of the configuration file and the current firmware version are the same.
Figure 2-5-3 Upload and Backup Configuration Files
Figure 2-5-4 Voice Record
Figure 2-5-5 Restore Configuration File
Figure 2-5-6 Restore System
Notice: You can restore the gateway to factory settings by dialing. Connect the phone to the FXS port of the gateway and dial "*1*2*3*4" , then it will restore the gateway to factory settings.
On the “Information” page, there shows some basic information about the analog gateway. You can see software and hardware version, storage usage, memory usage and some help information.
Figure 2-6-1 System Information Interface
3. Analog Settings
You can see much information about your ports on this page.
3.1 Channel Settings
Figure 3-1-1 Channel Settings
Click the button to automatically modify the corresponding port information.
Figure 3-1-2 FXO Port Configuration
Figure 3-1-3 FXS Port Configuration
3.2 Pickup Settings
Call pick-up is a feature used in a telephone system，which allows one phone to answer a call on the another phone. You can set the “Time Out” and “Number” parameters individually or globally for each port. This function is realized by dialing a series of specific numbers, provided that you enable this function and set the "number" parameter correctly.
Figure 3-2-1 Pickup Settings
Table 3-2-1 Definition of Pick-up
Set the timeout, in milliseconds (ms).
Notice: You can only enter numbers.
3.3 Dial Matching Table
The dial matching table is to effectively judge whether the received number is complete so that it can be sent in time.
The correct use of the dial matching table can help shorten the call establishment time.
Figure 3-3-1 Dial Matching Table
3.4 Global Settings
Figure 3-4-1 General Configuration
Table 3-4-1 Instruction of General
The duration of the signal tone generated by the corresponding channel. (in milliseconds).
The specified timeout period (seconds) for dialing a specific device.
Echo algorithm type
Default 100 (milliseconds)
Echo cancellation signal length
Software echo cancellation signal length
Default Loop start, busy tone is generated, Kewlstart, power is off, no busy tone is generated
Figure 3-4-2 Fax Configuration
Table 3-4-2 Definition of Fax Option
Set transfer mode
Set the sending and receiving baud rate.
Error correction mode
Turning on or off T.30 ECM (Error correction mode) is turned on by default.
Figure 3-4-3 Country Configuration
Table 3-4-3 Definition of Country Setting
Set the signal tone standard of the country where the gateway is located.
List of continuous ringing.
Set the off-hook dial tone.
Set the prompt tone to the caller when ringing.
Set the prompt tone when busy.
Call waiting tone
Set the background prompt tone to play when entering the call waiting.
Set the prompt tone to be played when congested.
Second dial tone
Set the prompt tone for the second dialing after pressing the flash key.
Set the prompt tone for the recording process.
Special message tone
Set the prompt tone for playing special information (for example: the dialed number is not in the service area).
3.5 Special Function Keys
Figure 3-5-1 Function keys
Figure 3-6-1 Calling ID
Table 3-6-1 Definition of Calling Number
How to send CID
Some countries (such as the United Kingdom) have different ring tone systems, which means that CID needs to be set later, not just after the first ring. And the default is after the first ring.
How long to wait before sending CID
The length of time (in milliseconds) we need to wait before sending the CID to the channel.
Turn on or off the hook/flash function.
Minimum hook time
The minimum hook-up time. (millisecond)
Maximum hook time
The maximum time for hooking. (millisecond)
"#" as the end of dialing
Turn the dial key on or off.
Show extension number
Set whether to display the extension number.
Figure 3-6-2 Other Parameters
Table 3-6-2 Definition of Other Parameters
The anti-jitter delay value when the gateway FXS port detects a hang-up signal. The set value is 32ms to 2048ms (multiples of 32), and the default value is 64ms.
Figure 3-6-2 Other Parameters
Figure 3-7-1 Regular
Table 3-7-1 Definition of Regular
Global encoding settings: mulaw, alaw
Figure 3-7-2 CallerID Detect
Table 3-7-2 Instructions of Calling Number Detection
Non-standard CID detection
Adapt to the detection function switch that is different from the standard CID
CID media stream length
CID media stream length byte size
CID media stream header length
CID media stream header length byte size
Delay of sending polarity signal
Delay time of sending polarity line reversal signal
Figure 3-7-3 Hardware Gain
Table 3-7-3 Instruction of Hardware gain
FXO to terminal gain
Set the gain of FXO to the terminal, the range of values that can be filled: from -150 to 120
FXO to IP gain
Set FXO to IP gain, the range of possible values: from -150 to 120
FXS to terminal gain
Set FXS to terminal gain, optional value: -35,0,35
FXS to IP gain
Set FXS to IP gain, optional values: -35,0,35
4.1 SIP terminal
On this page, the status information about the SIP account is displayed.
Figure 4-1-1 SIP Endpoints
Click the edit button to modify the corresponding SIP information.
4.1.1Main Endpoint Settings
There are 3 kinds of registration types for choose on the VoxStack series analog gateways. 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 Main Endpoint Settings
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 Main Endpoint Settings
When "Gateway is registered to the endpoint", you need to fill in the username and password, and you can register multiple SIP endpoints to the server. Due to the difference in usernames and passwords, there will be no confusion between routing and Trunks.
Figure 4-1-4 Main Endpoint Settings
Table 4-1-1 Definition of Endpoint Settings Options
A name which is able to read. And it’s only used for user’s reference.
Username for authentication between the endpoint and the gateway.
The password for authentication between the endpoint and the gateway, allowing letters.
Endpoint registers with this gateway---The gateway is used as a server, and the SIP endpoint is registered to the gateway;
This gateway registers with the endpoint---The gateway is used as a client, and the SIP terminal needs to be registered on the server.
Domain name or IP address
IP address or domain name of the endpoint or 'dynamic' .（if the endpoint has a dynamic IP address.）This needs to register.
Alternate domain name or IP address
Same as above.
After filling this, it is equivalent to that this account initiates registration to two domain names or IP addresses at the same time.
When the account of the primary domain name or IP address expires, it will switch to the account of the alternate domain name or IP address
Set possible transmission types and order of use for outgoing transmissions.
When you use various transport protocols： UDP, TCP, TLS，the transmission type enabled for the first time is only used for outgoing messages until registration occurs.
If the endpoint requires another transmission type during the registration process, the first transmission type may be changed to another transmission type.
Issues related to NAT addresses when incoming SIP or media sessions.
When the endpoint is a VOS server, the encryption item needs to be used, and the parameters need to be turned on at this time
4.1.2 Advanced: Registration Options
Figure 4-1-5 Registration Options
Table 4-1-2 Definition of Registration Options
A username to use only for registration.
When Gateway registers as a SIP user agent to a SIP proxy (provider), calls from this provider connect to this local extension.
Registered user name
The registered username，is the user in ”register => user[:secret[:authuser]]@host[:port][/extension]“
A username to identify the gateway to this endpoint.
A domain to identify the gateway to this endpoint.
The port number the gateway will connect to at this endpoint.
To check the endpoint's connection status whether or not.
How often, in seconds, to check the endpoint's connection status.
A proxy to which the gateway will send all outbound signaling instead of sending signaling directly to endpoints.
Customized registration switch
After opening, customers can customize the registration form by themselves.
Enable Outboundproxy to replace Host
4.1.3 Call Settings
Table 4-1-3 Definition of Call Options
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).
Set a call limit, the maximum number of calls that can be allowed at the same time.
Whether or not the Remote-Party-ID header should be trusted.
Whether or not to send the Remote-Party-ID header.
Endpoint 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.
4.1.4 Advanced: Signaling Settings
Table 4-1-4 Definition of Signaling Options
Inbound In-band Signaling
Whether to generate an incoming ring tone.
Never: indicates that the incoming signal is never applicable;
Optional values: yes, no, never;
Default value: never;
Allow Overlap Dialing
Allow Overlap Dialing: Whether or not to allow overlap dialing. Disabled by default.
Append User=Phone to URI
Whether or not to add ‘; user=phone’ to URIs that contain a valid phone number.
Add Q.850 Reason Headers
Whether or not to add Reasonable header and to use it if it is available.
SDP Version Header
By default, The gateway will add a session version number to the SDP packet and if the SDP version number is modified, it will only modify the SDP session.
Turning off this option will force the gateway to ignore this SDP version number and treat all SDP data as new data.
This is necessary for a device that sends non-standard SDP packets.
It is turned on by default.
Whether or not to globally enable transfers. Choosing 'no' will disable all transfers (unless enabled in endpoints or users). Default is enabled.
Allow Promiscuous Redirection
Whether or not to allow 302 or REDIR to non-local SIP address.
Notice：Redirecting to the local system will cause a loop call, which Asterisk does not support.
Setting for the SIP Max-Forwards header (loop prevention).
Send TRYING on REGISTER
Send a 100 Trying when the endpoint registers.
4.1.5 Advanced: Timer Settings
Table 4-1-5 Definition of Timer Options
Default T1 Timer
This timer is used primarily in INVITE transactions. The default for Timer
The default T1 clock is 500 milliseconds or if you have qualify=yes it will measure the round-trip time between the running gateway and the device.
Call Setup Timer
If no provisional response is received within this period of time, the call will be automatically blocked. The default value is 64*T1.
There are three modes to choose from: Proactively initiate，request and run the session timer;
Only accept or run the session timer when requested by other user agents;
Refuse, do not run session timers in any case.
Minimum Session Refresh Interval
The minimum session refresh interval. (in seconds) The default is 90secs.
Maximum Session Refresh Interval
The maximum session refresh interval.( in seconds) The default is 1800secs.
Session refresher, user agent client or user agent server. The default is the user agent server.
4.1.6 Media Settings
Table 4-1-6 Definition of Media Settings
Select codec from the drop down list. Different encoding priorities choose different encoding methods.
4.2 FXS Batch Binding SIP Accounts
If you want to bind SIP accounts in batches on the FXS port, you can configure this page.
Notice: It is only available in the "This gateway registers with the endpoint" working mode.
Figure 4-2-1 FXS Batch Binding SIP Accounts
4.3 Batch Create SIP Accounts
On this interface, users can create multiple SIP accounts at one time. You can choose any registration mode.
Figure 4-3-1 Batch Create SIP
4.4 Advanced SIP Settings
Table 4-4-1 Regular choice
UDP Bind Port
Choose a port on which to listen for UDP traffic.
Enable request server for incoming TCP link(default is no).
TCP Bind Port
Choose a port on which to listen for TCP traffic.
TCP Authentication Timeout
The maximum number of seconds for client link verification. If the client is not authenticated before the time expires, the client will be disconnected. (Default value: 30 seconds)
TCP Authentication Limit
The maximum number of simultaneous links allowed in a given time. (Default value: 50 seconds)
Enable Hostname Lookup
Open the DNS SRV lookup for outbound calls.
Notice: The gateway is only the first host in the SRV record. This function can be used in dial-up activation to dial SIP calls on the Internet through the domain name.
4.4.2 NAT Settings
Table 4-4-2 Definition of NAT Settings
Format:192.168.0.0/255.255.0.0 or 172.16.0.0./12.
A list of IP address or IP ranges which are located inside a NAT network.
This gateway will replace the internal IP address in SIP and SDP messages with the external IP address when a NAT exists between the gateway and other endpoints.
Local Network List
Local IP address list that you added.
Subscribe Network Change Event
Through the use of the “test_stun_monitor” module, the gateway has the ability to detect when the perceived external network address has changed. When the “stun_monitor” is installed and configured, “chan_sip” will renew all outbound registrations when the monitor detects any sort of network change has occurred. By default this option is enabled, but only takes effect once “res_stun_monitor” is configured. If “res_stun_monitor” is enabled and you wish to not generate all outbound registrations on a network change, use the option below to disable this feature.
Match External Address Locally
Only substitute the external address or domain name if it matches.
Dynamic and Static Selection
Dynamic hosts are not allowed to register with the IP address of static hosts. This will avoid registration errors with the same IP.
External TCP Port Mapping
When the gateway is behind a static NAT or PAT, the TCP port is externally mapped.
External IP Address
The external address of the NAT. (and optional TCP port)
External IP Address = hostname[:port] specifies a static address[:port] to be used in SIP and SDP messages.
External IP Address = 220.127.116.11
External IP Address = 18.104.22.168:9900
External IP Hostname
The external hostname (and optional TCP port) of the NAT.
Hostname Refresh Interval
It will show how often to perform the hostname lookup. You can also configure a domain name. The gateway will perform a DNS query. (This method is not recommended). Try to use IP and configure “externip”.
4.4.3 STUN Settings
Table 4-4-3 Definition of STUN Settings Options
Turn on function
Default port 3478
Refresh Request Interval
Time interval in seconds, default 30 seconds
Server IP Address/Domain Name
Server address or domain name
4.4.4 RTP Settings
Table 4-4-4 Definition of RTP Settings Options
Start of RTP Port
Start range of port numbers to be used for RTP.
End of RTP port
End range of port numbers to be used for RTP.
4.4.5 Parsing and Compatibility
Table 4-4-5 Instruction of Parsing and Compatibility
Strict RFC Interpretation
Check header tags, character conversion in URIs, and multiline headers for strict SIP compatibility(default is yes)
When speaking back to other SIP peers, the other peers should include an “Allow” header to tell us the implementation of the SIP method. However, some peers do not include “Allow” headers or forge the methods they implement. In this case, the gateway will assume that the peer supports all known SIP methods. If you know that your SIP peer does not provide support for a specific method, then you may need to provide a list of methods that the peer does not implement in “disallowed methods”.
Notice: If your peer is real, then there is no need to set this option
Shrink Caller ID
The function can 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 allowable time for incoming registration and subscription.(seconds)
Minimum Registration Expiry
The minimum length of registration and subscription.(default 60)
Default Registration Expiry
Default length of incoming/outgoing registration.
How long will it take to re-register the extension. (Default 20 seconds)
Number of registration attempts
Number of registration attempts before giving up.
4.4.6 Security and Media
Table 4-4-6 Instruction of Security and Media
Match Auth Username
If available, Use the user name field of the authentication line to match instead of using the user name of the user name field.
For authentication domains, all domains must be globally unique according to the RFC3261 standard. Generally can be set to host name or domain name.
Use Domain as Realm
Use the SIP domain as the boundary of the domain.
Always Auth Reject
When an “INVITE” or “REGISTER” request is rejected for any reason, the same reason will always be used. The username is legal but the password is incorrect. It does not tell the requester whether there is this “user” or “peer”, which reduces the possibility of an attacker scanning the SIP account. (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 customer calls (enabled by default, allowed). If your gateway is connected to an external network and allows customers to call, you want to check which services are provided for everyone and enable it in the default “context”.
Some SDN links will 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 medias 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 'no answer' applications to work, you need to run the progress() application in the priority before the app.
TOS for SIP Packets
Sets type of service for SIP packets
TOS for RTP Packets
Sets type of service for RTP packets
4.5 Sip Account Security
This analog gateway support TLS protocl for encrypting calls. On the one hand, it can worked as TLS server, generate the session keys used for the secure connection. On the other hand, it also can be registered as a client, upload the key files provied by the server.
Figure 4-5-1 TLS Settings
Table 4-5-1 Instruction of TLS
Enable or disable DTLS-SRTP support.
TLS Verify Server
Enable or disable TLS verify server(default is no).
Specify the port for remote connection.
TLS Client Method
Values include tlsv1, sslv3, sslv2, Specify protocol for outbound client connections (default is sslv2)
The gateway has a friendly user interface and very flexible settings. It supports up to 512 routing rules and each routing rule supports up to 100 pairs of calling/called number filtering and conversion operations. It supports DID function (the use of DID function: how to use the T1/E1 gateway DID function of China Telecom). The gateway supports trunk group and trunk priority management
5.1 Call Routing Rules
Figure 5-1-1 Routing Rules
Click “Add”,you can set up a new routing rule. Click "Edit" to modify the routing rule, and click "Delete" to delete the routing rule.
Figure 5-1-2 Example of Setup Routing Rule
Table 5-1-1 Definition of Call Routing Rule
This is a rule name. The type of match usually used to describe (for example, ‘sip1 TO port1’ or ‘port1 TO sip1’).
Call Is From
Source of the call.
The destination to receive the incoming calls.
The specific setting time of DISA timeout.
Maximum Number of Digits In Password
Set the maximum number of password digits
Set a password within the specified range
Figure 5-1-3 Advanced Routing Rule
Table 5-1-2 Definition of Advanced Routing Rule
Calling/called Number Filtering And Conversion
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).
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)
*matches one or more digits
Prepend<add prefix>: The number added when the pattern matches successfully. If the dialed number matches the pattern specified in the subsequent column, the number will be added before being sent to the trunk.
Prefix: Removed when the pattern is matched successfully. The dialed number is matched with the pattern specified in the subsequent column. Once the match is successful, the prefix will be removed from the number before being sent to the trunk.
Match Pattern: The dialed number will be compared with the number in the ”prefix +” this matching pattern. Once the match is successful, the matched pattern part of the dial will be sent to the trunks.
SDfR<Delete digits from the right>: The number of digits to be deleted from the right end of the number. If this value of this item exceeds the length of the current number, the entire number will be deleted.
RDfR<Reserved digits from the right>: The reserved digits from the right.
StA<Add Suffix>: Add this number from the right end of the current number.
Caller Name <caller display name>: Set your favorite caller name before sending this call to the terminal, allowing the use of local languages, such as Chinese and Latin.
Time Patterns that will use this Route
Time mode setting of routing rules.
What destination number will you dial?
This is very useful when you have a transfer call.
Call failure handling
The gateway will attempt to send the call out each of these in the order you specify.
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 Ports or SIP to groups. Then if you want to make a call, it will find available port automatically.
Figure 5-2-1 Group Rules
You can click the "Add" button to set up a new group, if you want to modify an existing group, you can click the "Edit" button
Figure 5-2-2 Create a Group
Figure 5-2-3 Modify a Group
Table 5-2-1 Definition of Routing Groups
The name of the route, used to describe the type of this call route, for example, sip1 TO port1 or port1 TO sip2.
5.3 Batch Create Rules
If you bind telephone for each FXO port and want to establish separate call routings for them. For convenience, you can create call routing rules for each FXO port at once in this page in batches.
Figure 5-3-1 Batch Create Rules
6.1 Network Settings
There are three types of LAN port IP to choose from：Factory, Static and DHCP. The default type is: factory, the default IP is 172.16.99.1. If you forget the current IP, you can connect the phone to any FXS port of the analog gateway and dial "**" to query the current IP.
Figure 6-1-1 LAN Settings Interface
Table 6-1-1 Definition of Network Settings
The name of network interface.
The method to get IP.
Static: manually set up your gateway IP.
DHCP: dynamically obtain the gateway IP address.
The physical address of the network interface.
The IP address of your gateway.
The subnet mask of your gateway.
Default getaway IP address.
Reserved Access IP
List of domain name server IP addresses. This information is mainly obtained from the local network service provider.
Enable or disable the reserved IP address switch.
The reserved IP address for this gateway.
The subnet mask of the reserved IP address.
6.2 VPN Settings
You can select VPN type and upload OpenVPN client configuration file or fill in PPTP VPN account information. If successful, you can see a VPN virtual network card on the system status page. You can refer to the parameter hints and sample configuration.
Figure 6-2-1 VPN Interface
6.3 DDNS Settings
You can enable or disable DDNS (Dynamic Domain Name Server) according to your needs
Figure 6-3-1 DDNS Interface
Table 6-3-1 Definition of DDNS Settings
Enable/Disable DDNS(dynamic domain name server)
Set the type of DDNS server.
Your DDNS account’s login name.
Your DDNS account’s password.
The domain to which your web server will belong.
This tool is used to detect the network connection, you can execute the Ping command on the web interface
Figure 6-4-1 Network Connectivity Checking
Figure 6-4-2 Channel Recording
Figure 6-4-3 Capture Network Data
Table 6-4-1 Definition of Channel Recording
The name of network interface.
Source Host Address
Specify the source address of the data you want to get
Specify the destination address you want to get data from
Specify the port where you want to get data
Specify the channel number you want to get data
Tcpdump Option Parameter
The tool of tcpdump capture network data by parameter option specified.
6.5 Security Settings
Figure 6-5-1 Security Settings Interface
6.6 Firewall Security Rules
Figure 6-6-1 Firewall Security Rules Interface
7.1 Asterisk API
When you make “Enable” switch to “on”, this page is available.