FGX4508 Admin Guide

VoIP GSM Gateway #

Confidentiality
The information contained in this document is highly confidential and proprietary to FIBERME Communications LLC. It may not be distributed, reproduced, or disclosed, either orally or in writing, to any third party without the prior written consent of FIBERME Communications LLC.

Disclaimer
FIBERME Communications LLC reserves the right to change the design, features, and products at any time without prior notice or obligation. The company shall not be held responsible for any errors or damages of any kind arising from the use of this document.

FIBERME has made every effort to ensure the accuracy and completeness of the information in this document. However, the contents are subject to change without prior notice. Please contact FIBERME to verify that you have the most up-to-date version of this document.

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



1. Overview #

1.1 What is FGX4508? #

The FIBERME FGX4508 series gateways, which include the FGX4508G and FGX4508L models, support multiple codecs such as G.711U, G.711A, GSM, G.722, G.726, and G.729. These gateways offer features like SMS sending, receiving, group messaging, and SMS-to-email functionality. The FGX4508 series is fully compatible with FIBERME FCM, Asterisk, 3CX, FreePBX, FreeSWITCH, Cisco, AVAYA SIP servers, and the VOS VoIP platform. These products are designed to help users significantly reduce telecommunications and communication costs.


1.2 Product Introduction #

The FGX4508 series gateways are available in a variety of models, and each model supports a different number of ports and frequency bands. The following table shows:

ModelModulePortsNetwork InterfaceBand
FGX4508GGSM82850/900/1800/1900MHz@GSM
FGX4508LLTE82LTE FDD: B1/B3/B5/B8
LTE TDD: B38/B38/B40/B41
WCDMA: B1/B8
TD-SCDMA: B34/B39
CDMA: BC0
GSM: 900/1800MHz

1.3 Application #

1.3.1 LCD And Buttons #

LED Indicator/Icon/ButtonsColor/IconStatus
Network Status LEDGreen and FlashNetwork Connected
RUN LEDAlways GreenThe system is starting
  Green and FlashThe system runs
ALM LEDAlways RedThe system is starting
  LED is offThe system runs
  Red and FlashAsterisk exception
Port LEDLED is offNo sim card inserted or port is not available
  Green and FlashSim card is in use
  Always GreenSim card is available
RST ButtonPress and hold the RST button for 3-5 seconds. The display jumps to the “System Booting” page to restart the system.

  #

1.3.2 Product Panel #

Front panel of FGX4508

Rear Panel of FGX4508


1.4 Main Features #

  • Wide selection of codecs and signaling protocol
  • Support SMS sending, receiving, group sending
  • Support transferring SMS to E-mail
  • Support SMS remotely controlling gateway
  • Support USSD service
  • Support PIN identification
  • Support unlimited routing rules and flexible routing settings
  • SIM cards are all hot-swap
  • Stable performance, flexible dialing, friendly GUI

1.5 Physical Information #

  • Size(No antenna and hanging ears): 440mm*44mm*300mm
  • LAN port:1
  • WAN port:1
  • USB Interface:1
  • SIM Cards: hot-swap
  • Operation Temperature: 0~45°C
  • Storage Temperature: -20~70°C
  • Operation humidity:10% ~ 90% non-condensing

1.6 Software #

  • Default IP:172.16.98.1
  • Username: admin
  • Password: admin

For initial access, you can connect to the FGX4508 using the default IP address 172.16.98.1. From there, you can configure the module according to your requirements.



2. System #

2.1 Status #

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

Description of System Status

OptionsDefinition
PortNumber of each ports.
SignalDisplay the signal strength of in each channels of gateway.
BERBit Error Rate.
CarrierDisplay the network carrier of current SIM card.
Registration StatusIndicates the registration status of current module.
PDDPost Dial Delay (PDD) is experienced by the originating customer as the time from the sending of the final dialed digit to the point at which they hear ring tone or other in-band information. Where the originating network is required to play an announcement before completing the call then this definition of PDD excludes the duration of such announcements.
ACDThe Average Call Duration (ACD) is calculated by taking the sum of billable seconds (bill sec) of answered calls and dividing it by the number of these answered calls.
ASRAnswer Seizure Ratio is a measure of network quality. Its calculated by taking the number of successfully answered calls and dividing by the total number of calls attempted. Since busy signals and other rejections by the called number count as call failures, the ASR value can vary depending on user behavior. ModuleStatus Show the status of port, include blank space and “READY”. Black space means it is unavailable here and “Ready” means the port is available
Module StatusDisplay the status of the port. “Ready” means registering and “READY” means port is available
Remain TimeThis value is multiplied by to step length is a rest call time.

2.2 Time #

Description of Time Settings

OptionsDefinition
System TimeYour gateway system time
Time ZoneThe world time zone. Please select the one which is the same or the closest as your city
POSIX TZ StringPosix time zone strings.
NTP Server 1Time server domain or hostname. For example,
[time.asia.apple.com].
NTP Server 2The first reserved NTP server. For example, [time.windows.com].
NTP Server 3The second reserved NTP server. For example, [time.nist.gov].
Save DataSave the Modify of the time settings
Sync from NTPSync time from NTP server.
Sync from ClientSync time from local machine.

For example, you can configure like this:

Time Settings


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


2.3 Login Settings #

You can modify “Web Login Settings” and “SSH Login Settings”. If you have changed these settings, you don’t need to log out, just rewriting your new user name and password will be OK. Also you can specify the web server port number. Normally, the default web login mode is “http and https.” For security, you can switch to “only https”.

Description of Login Settings

OptionsDefinition
User NameDefine your username and password to manage your gateway
Allowed characters “-_+. < >&0-9a-zA-Z”. Length: 1-32 characters.
PasswordAllowed characters “-_+. < >&0-9a-zA-Z”. Length: 4-32 characters.
Confirm PasswordPlease input the same password as ‘Password’ above.
Login Modehttp and https: You can access gateway via link: http://gatewayIP or https://gatewayIP
https: You can only access gateway via link: https://gatewayIP
PortSpecify the web server port number.

For example, you can configure as below:

Login Settings

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


2.4 General #

2.4.1 Language Settings #

You can choose different languages for your system. If you want to change language, you can switch “Advanced” on, then “Download” your current language package. After that, you can modify the package with the language you need. Then upload your modified packages, “Choose File” and “Add”.
For example:


Language Settings

2.4.2 Scheduled Reboot #

If switch it on, you can manage your gateway to reboot automatically as you like. There are four reboot types for you to choose, “By Day, By Week, By Month and By Running Time”.

Reboot Type

If use your system frequently, you can set this enable, it can helps system work more efficient.


2.5 Tools and Information #

2.5.1 Reboot Tools #

You can choose system reboot and asterisk reboot separatel

Reboot Tools

If you press “OK”, your system will reboot and all current calls will be dropped. Asterisk Reboot is the same.

2.5.2 Update Firmware #

We offer 2 kinds of update types for you; you can choose System Update or System Online Update. If you choose System Online Update, you will see the following information:


Update Firmware

2.5.3 Upload and Backup Configuration #

If you want to update your system and remain your previous configuration, you can first backup configuration, then you can upload configuration directly. That will be very convenient for you.

Upload and Backup Configuration


2.5.4 Restore Configuration #

Sometimes there is something wrong with your gateway that you don’t know how to solve it, mostly you will select factory reset. Then you just need to press a button, your gateway will be reset to the factory status.

Restore Configuration


2.6 Information #

On the “Information” page, there shows some basic information about the gateway. You can see software and hardware version, storage usage, memory usage and some help information.

Information



2.7 User #

On the “User” page, webpage accounts can be added via admin user. You can add different accounts with different rights.

Add user



3. PORTS #

3.1 Ports Settings #

Ports Settings

On this page, you can see your SIM Card information and module status, click action button to configure the port.


Port Configuration


If you have set your Pin Code, you can check on like this:

PIN Code Application

If you want to hide your number when you call out, you can just switch CLIR “ON” (Of course you need your operator’s support)


CLIR Application

Definition of Module Settings

OptionsDefinition
NameThe alias of each port. Input name without space here.
Allowed characters “-_+.<>&0-9a-zA-Z”.Length: 1-32 characters.
Speaker VolumeThe speaker volume level, the range is 0-100.
This will adjust the loud speaker volume level by an AT command.
Microphone VolumeThe microphone volume, range is: 0-15.
This will change the microphone gain level by an AT command.
Dial PrefixThe prefix number of outgoing calls from this channel
PIN Code Personal identification numbers of SIM card. PIN code can be modified to prevent SIM card from being stolen.
Custom AT commands when startUser custom AT commands when start system, use  “|” to split AT command
CLIR Caller ID restriction, this function is used to hidden caller ID of SIM card number. The gateway will add ‘#31#’ in front of mobile number. This function must support by Operator.
SMS Center NumberYour SMS center number of your local carrier.
Module IMEIOnly CDMA module does not support modifying IMEI

3.2 Advanced #

Let device register EPS network. Note: only for 4G or above.

Advanced


3.3 Call Forwarding #

You can set call forwarding unconditional, no reply, busy and unreachable.

Call Forwarding


3.4 Call Waiting #

You can open, close or query call waiting here.

Call Waiting


3.5 DTMF #

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


DTMF Detection Settings

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

OptionsDefinition
DTMF Normal Twist and Reverse TwistIt 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 RowThe 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 ColThe 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 BeginSampling matching value. You can choose 2 or 3.
DTMF Misses EndThe time interval between the two digits you input. Adjust the speed of input. The smaller value represents the shorter intervals.

Description of DTMF Detection Settings


3.6 Toolkit #

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

Function Options

 Description of Definition of Functions

OptionsDefinition
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 USSDEnter a specific USSD number (For example,*142# to check your SIM card’s balance. This USSD number is might be different from different carriers) to get the USSD information. The gateway will try to get by AT commands.
AT CommandTo perform some specific AT commands. This is useful when you have a debug of the modem. e.g. perform [ AT+CSQ ] to check what signal qualify it is. In AT commands, there is no difference between “a” and “A”

If you want to send AT command, first you should input your command, then select certain ports and choose “Copy to Selected“, finally choose “Execute“.

AT Command Example


3.7 Ports Update #

Update port and MCU firmware.

 Port Update


3.8 Call and SMS limit #

all and SMS Limit

We offer you Call Limit, Call Time Limit, Lock Sim, SMS limt.



3.8.1 Call Limit #

Call Limit

We offer limit number of outbound calls per day, limit number of inbound calls per day and limit number of outbound calls per hour.

3.8.2 Call Limit Time #

Now we can offer you two types of call duration limit, you can choose “Single Call Duration Limit” or “Call Duration Limitation” to control your calling time

Single Call Duration Limit: This will limit the time of each call.
First you need to switch “Enable” on, then you can set “Step” and “Single Call Duration Limitation” any digits you want. When you make a call by this port, it will limit your calling time within the product of

Step * Single Call Duration Limitation
And if your calling time overtops the value above, the system will hang up this call.

Single Settings

Call Duration Limitation:  This will limit your total calling time of this port. If remain time is 0, it will not send calls through this port.


 Call Duration Limitation Settings


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


 Auto Reset Settings

Description of Call Duration Limit Settings

OptionsDefinition
StepStep length value range is 1-999s, step length multiplied by time of single call just said a single call duration time allowed.
Enable Single Call Duration LimitDefinite 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 LimitationThis function is to limit the total call duration of channel. The max call duration is between 1 to 999999 minutes.
Minimum Charging TimeA single call over this time, Module side of the operators began to collect fees, unit for seconds.
Alarm Threshold Define a threshold value of call minutes, while the call minutes less than this value, the gateway will send alarm information to designated phone.
Alarm Description Alarm port information description, which will be sent to user mobile phone with alarm information.
Alarm Phone NumberReceiving alarm phone number, user will received alarm message from gateway.
Enable Auto ResetAutomatic restore remaining talk time, that is, get total call minutes of each channel.
Auto Reset TypeReset 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

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3.8.3 SMS Limit #

You can limit the number of SMS messages sent per day or per month.

 SMS Limit

You can save your configuration to other ports.


Save to Other Ports

If you have set like this, you will see many IMG_289 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.

Port Information



4. VOIP #

4.1 VOIP Endpoints #

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

SIP&IAX2 Endpoints

4.1.1 Add New SIP Endpoint #

Main SIP Endpoint Settings:
You can click Add New SIP Endpoint button to add a new SIP endpoint, and if you want to modify existed endpoints, you can click EDIT button.
There are 3 kinds of registration types for choose.  None, Server or Client.
You can configure as follows:
If you set up a SIP endpoint by registration “None” to a server, then you can’t register other SIP endpoints to this server. (If you add other SIP endpoints, this will cause Out-band Routes and Trunks confused.)

None Registration

For convenience, we have designed a method that you can register your SIP endpoint to your gateway, thus your gateway just works as a server.

Server

Also you can choose registration by “This gateway registers with the endpoint”, it’s the same with “None”, except name and password.

Client

Definition of SIP Options

OptionsDefinition
NameDisplay name
UsernameRegister name in your SIP server
PasswordAuthenticating with the gateway and characters are allowed.
RegistrationNone — Not registering;    Server — When register as this type, it means the gateway acts as a SIP server, and SIP endpoints register to the gateway;   Client — When register as this type, it means the gateway acts as a client, and the endpoint should be register to a SIP server;
Hostname or IP AddressIP address or hostname of the endpoint or ‘dynamic’ if the endpoint has a dynamic IP address. This will require registration.
TransportThis 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 TraversalNo — Use Report if the remote side says to use it. Force Report on — Force Report to always be on. Yes — Force Report to always be on and perform comedia RTP handling. Report if requested and comedia — Use Report if the remote side says to use it and perform comedia RTP handling.

Advanced——Registration Options

Advanced Registration Options

Definition of Registration Options

OptionsDefinition
Authentication UserA username to use only for registration.
Register ExtensionWhen Gateway registers as a SIP user agent to a SIP proxy (provider), calls from this provider connect to this local extension.
Register UserRegister user name , it is the user of register => user[:secret[:authuser]]@host[:port][/extension]
Contact UserWhen the Contact User is 402 Contact: <sip:402@172.16.6.123:5060;transport=UDP
From UserA username to identify the gateway to this endpoint.
From DomainA domain to identify the gateway to this endpoint.
QualifyWhether or not to check the endpoint’s connection status
Qualify FrequencyHow often, in seconds, to check the endpoint’s connection status.
Outbound ProxyA proxy to which the gateway will send all outbound signaling instead of sending signaling directly to endpoints.

Call Settings

Call Settings

Definition of Call Options

OptionsDefinition
DTMF ModeSet 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-IDWhether or not the Remote-Party-ID header should be trusted.
Send Remote-Party-IDWhether or not to send the Remote-Party-ID header.
Caller ID PresentationWhether or not to display Caller ID.
Call LimitUsually used when this sip work as a trunk. To limit number of maximum channels supported by the sip trunk.

Advanced:——Signaling Settings


Signaling Settings

Definition of Signaling Options

OptionsDefinition
Progress InbandWhether there is ringing tone.
Never: Indicates that incoming calls are never applicable.
Optional values: yes / no / never. Default: yes
Append user=phone to URI Whether or not to Add ‘user = phone’ to UPIS to include a valid phone number in the URI.
Add Q.850 Reason HeadersIf it is available, Whether or not to add a reason header and use it.
Honor SDP VersionWhether or not to display Caller ID.
Allow TransfersWhether or not to globally enable transfers. Choosing ‘no’ will disable all transfers (unless enabled in peers or users). Default is enabled.
Allow Promiscuous RedirectsWhether or not to allow 302 or REDIR to non-local SIP address. Note that promiscredir when redirects are made to the local system will cause loops since this gateway is incapable of performing a “hairpin” call.
Max Forwards Setting for the SIP Max-Forwards header (loop prevention). Send TRYING on REGISTER Send a 100 Trying when the endpoint registers.
Send TRYING on RegisterWhether send a 100 Trying when the endpoint registers

Advanced——Timer Settings

Timer Settings

Definition of Timer Options

OptionsDefinition
Default T1 TimerThis 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.
Default T2 TimerThis timer is used primarily in INVITE transactions. The default for Timer T2 is 4000 ms 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 TimersSession-Timers feature operates in the following three modes: originate, Request and run session-timers always; accept, run session-timers only when requested by other UA; refuse, do not run session timers in any case.
Minimum Session Minimum session refresh interval in seconds. Default is 90secs.
Maximum Session Refresh IntervalMaximum session refresh interval in seconds. Defaults to 1800secs.
Session RefresherThe session refresher, UAC or UAS. Defaults to UAS.

 4.1.2 Add New IAX2 Endpoint #

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

None Registration

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


Server

Also, you can choose registration by “This gateway registers with the endpoint”, it will work as a client.


Client

Definition of IAX2 Options

OptionsDefinition
NameDisplay name
UsernameAuthentication name in your IAX2 server
PasswordAuthenticating with the gateway and characters are allowed.
RegistrationNone — Not registering; Endpoint registers with this gateway — When register as this type, it means the gateway acts as a IAX2 server, and IAX2 endpoints register to the gateway; This gateway registers with the endpoint — When register as this type, it means the gateway acts as a IAX2 client, and the endpoint should be register to a IAX2 server;
Hostname or
IP Address
IP address or hostname of the endpoint or ‘dynamic’ if the endpoint has a dynamic IP address. This will require registration.
AuthThere are three authentication methods that are supported: md5, plaintext and rsa. The least secure is “plaintext”, which sends passwords clear text across the net. “md5” uses a challenge/response md5 sum arrangement, but still requires both ends have plain text access to the secret. “rsa” allows unidirectional secret knowledge through public/private keys.If “rsa” authentication is used, “inkeys” is a list of acceptable public keys on the local system that can be used to authenticate the remote peer, separated by the “:” character. “outkey” is a single, private key to use to authenticate to the other side.
Transfer This application allows you to transfer calls.
Trunk“trunk=yes” Purpose: To obtain a better chart of actual bandwidth usage per codec as seen “on-the-wire” when using IAX2 trunking between two Asterisk telephony servers.


Advanced——Registration Options


Registration Options

Definition of Registration Options

OptionsDefinition
QualifyThe qualify settings are used to determine the status availability of an IAX peer. If a peer is considered to be in a reachable (OK or LAGGED) state, it is queried for availability every “qualifyfreqok” milliseconds. If it is considered to be in an UNREACHABLE state, it is queried for availability every “qualifyfreqnotok” milliseconds.The qualify= setting turns the qualify system on (if the “yes” or xxx options are used) or off (if qualify=no, which is by default). The millisecond value of the qualify= setting specifies the maximum response time of the availability acknowledgement before the peer is considered to be in a “LAGGED” state.
Qualify SmothingUse an average of the last two PONG result to reduce falsely detected LAGGED host. The default is ‘no’.
Qualify Freq OkHow frequently to ping the peer when everything seems to be OK, in milliseconds.
Qualify Freq Not OkHow frequently to ping the peer when it is either, LAGGED or UNAVAILABLE, in milliseconds.
Port The port number the gateway will connect to at this endpoint.

IAX2 Encryption


IAX2 Encryption


Definition of Encryption Options

OptionsDefinition
EncryptionEnable IAX2 encryption. The default is no.
Force EncryptionForce encryption insures no connection is established unless both sides support encryption. By turning this option on, encryption is automatically; turned on as well. The default is no

IAX2 Trunk Setting


IAX2Trunk Settings

Definition of Trunk Options

OptionsDefinition
Trunk Max Size Defaults to 128000 bytes, which supports up to 800; calls of ulaw at 20ms a frame.
Trunk MTUWith a large amount of traffic on IAX2 trunk, there is a risk of bad voice quality when allowing the Linux system to handle fragmentation of UDP packets. Depending on the side of each payload, allowing the OS to handle fragmentation may not be very efficient. This setting sets the maximum transmission unit for AIX2 UDP trunking. The default is 1240 bytes which means if a trunk’s payload is over 1240 bytes for every 20ms it will be broken into multiple 1240 bytes messages. Zero disables this functionality and let’s the OS handle fragmentation.
Trunk FrequencyHow frequently to send trunk msgs (in ms). This is 20ms by default.
Trunk Time StampsShould we send timestamps for the individual subframes within trunk frames? There is a small bandwidth use for these (less than 1kbps/call), but they ensure that frame timestamps get sent end-to-end properly. If both ends of all your trunks go directly to TDM, _and your trunkfreq equals the frame length for your codecs, you can probably suppress these. The receiver must also need to have it enabled.
Min. RegExpireMinimum amounts of time that IAX2 peers can request as a registration interval (in seconds).
Max. RegExpireMaximum amounts of time that IAX2 peers can request as a registration expiration interval (in seconds).

4.2 Batch SIP Endpoints #

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

Multiple SIP Extensions Settings

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


Definition of Multiple SIP Extensions

OptionsDefinition
NameDisplay name
UsernameRegister name in your SIP server
PasswordAuthenticating with the gateway and characters are allowed.
RegistrationNone — Not registering;    Server — When register as this type, it means the gateway acts as a SIP server, and SIP endpoints register to the gateway;   Client — When register as this type, it means the gateway acts as a client, and the endpoint should be register to a SIP server;
Hostname or IP AddressIP address or hostname of the endpoint or ‘dynamic’ if the endpoint has a dynamic IP address. This will require registration.
AutoPasswordTick – Automatically increments based on the password entered in the first line Do not check – All SIP endpoints have the same password as the first one.

4.3 Advanced SIP Settings #

4.3.1 Networking #

Networking General


Networking General

Definition of Networking General Optiongs

OptionsDefinition
UDP Bind PortUDP Bind Port
Enable TCPEnable server for incoming TCP connection (default is no).
TCP Bind PortChoose a port on which to listen for TCP traffic.
TCP Authentication TimeoutThe maximum number of seconds a client has to authenticate. If the client does not authenticate before this timeout expires, the client will be disconnected. (default value is: 30 seconds).
TCP Authentication LimitThe maximum number of unauthenticated sessions that will be allowed to connect at any given time (default is: 50).
Enable Hostname LookupEnable 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 CallWhether enable the internal SIP calls or not when you select the registration option “Endpoint registers with this gateway”.
Internal SIP Call PrefixSpecify a prefix before routing the internal calls.

NAT Settings


NAT Settings

Definition of NAT Settings Options

OptionsDefinition
Local NetworkFormat: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 ListLocal IP address list that you added.
Subscribe Network Change EventThrough 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 LocallyOnly substitute the extern addr or extern host setting if it matches.
Dynamic Exclude StaticDisallow 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 PortThe externally mapped TCP port, when the gateway is behind a static NAT or PAT.
External AddressThe external Address
External HostnameThe external hostname (and optional TCP port) of the NAT.
Hostname Refresh IntervalHow often to perform a hostname lookup. This can be useful when your NAT device lets you choose the port mapping, but the IP address is dynamic. Beware, you might suffer from service disruption when the name server resolution fails.

RTP Settings


RTP Settings

Definition of RTP Settings Options

OptionsDefinition
Start of RTP Port Range Start of range of port numbers to be used for RTP
End of RTP port RangeEnd of port numbers to be used for RTP
RTP TimeoutRTP Timeout re-transmission time

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4.3.2 Parsing and Compatibility #

Parsing and Compatibility

Instruction of Parsing and Compatibility

OptionsDefinition
Strict RFC InterpretationCheck header tags, character conversion in URIs, and multiline headers for strict SIP compatibility(default is yes)
Send Compact HeadersSend compact SIP headers
SDP OwnerAllows you to change the username filed in the SDP owner string. This filed MUST NOT contain spaces.
Ring 183 ModeImmediately or after ring
Disallowed SIP MethodsThe external hostname (and optional TCP port) of the NAT.
Shrink Caller IDThe 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 ExpiryMaximum allowed time of incoming registrations and subscriptions (seconds).
Minimum Registration ExpiryMinimum length of registrations/subscriptions (default 60).
Default Registration ExpiryDefault length of incoming/outgoing registration.
Registration TimeoutHow often, in seconds, to retry registration calls. Default 20 seconds.
Number of RegistrationAttempts Enter ‘0’ for unlimited Number of registration attempts before we give up. 0 = continue forever, hammering the other server until it accepts the registration. Default is 0 tries, continue forever.

4.3.3 Security #

Security Settings

Instruction of Security

OptionsDefinition
Match Auth UsernameIf available, match user entry using the ‘username’ field from the authentication line instead of the ‘from’ field.
RealmRealm for digest authentication. Realms MUST be globally unique according to RFC 3261. Set this to your host name or domain name.
Use Domain as RealmUse 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 RejectWhen 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 RequestsEnabling this option will authenticate OPTIONS requests just like INVITE requests are. By default this option is disabled.
Allow Guest CallingAllow or reject guest calls (default is yes, to allow). If your gateway is connected to the Internet and you allow guest calls, you want to check which services you offer everyone out there, by enabling them in the default context.

4.3.4 Media #

Media Settings

Instruction of Media

OptionsDefinition
Premature MediaSome ISDN links send empty media frames before the call is in ringing or progress state. The SIP channel will then send 183 indicating early media which will be empty – thus users get no ring signal. Setting this to “yes” will stop any media before we have call progress (meaning the SIP channel will not send 183 Session Progress for early media). Default is ‘yes’. Also make sure that the SIP peer is configured with progressinband=never. In order for ‘noanswer’ applications to work, you need to run the progress() application in the priority before the app.
TOS for SIP PacketsSets type of service for SIP packets
TOS for RTP PacketsSets type of service for RTP packets

  #

4.3.5 Codec Settings #

Select codecs from the list below.


Codec Settings


4.4 Advanced IAX2 Settings #

4.4.1 General Settings #

General Settings

Table 4-17 Instruction of General

OptionsDefinition
Bind PortBind port and bindaddr may be specified
Bind AddressMore than once to bind to multiple addresses, but the first will be the default.
Enable IAXCompatMore than once to bind to multiple addresses, but the first will be the default.
Enable No checksumsSet iaxcompat to yes if you plan to use layered switches or some other scenario which may cause some delay when doing a lookup in the dialplan. It incurs a small performance hit to enable it. This option cause Asterisk to spawn a separate thread when it receives an IAX DPREQ (Dialplan Request) instead of blocking while it waits for a response.
Enable Delay RejectDisable UDP checksums (if no checksums is set, then no checksums will be calculated/checked on system supporting the feature)
ADSIADSI (Analog Display Services Interface) can be enable if you have (or may have) ADSI compatible CPE equipment.
SRV LoopupWhether or not to perform an SRV lookup on outbound calls
AMA FlagsYou may specify a global default AMA flag for iaxtel calls. These flags are used in the generation of call detail records.
autokillIf we don’t get ACK to our NEW within 2000ms,and autokill is set to yes, then we cancel the whole thing(that’s enough time for one retransmission only ).This is used to keep things from stalling for a long time for a host that is not available for bad connections.
LanguageYou 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 CodeYou may specify a default account for Call Detail Records (CDRs) in addition specifying on a per-user basis.

4.4.2 Music on Hold #

Music on Hold Settings

Instruction of Music on Hold

OptionsDefinition
MohsuggestThe ‘Mohsuggest’ option specifies which music on hold class to suggest to the peer channel when this channel place the peer on hold. It may be specified globally or on a per-user or per-peer basis.
MohinterpretYou may specify a global default language for users. This can be specified also on a per-user basis. If omitted, will fall back to English(en)

  #

4.4.3 Instruction of Codec Settings #

Codec Settings

Instruction of Codec Settings

OptionsDefinition
Band WidthSpecify bandwith of low, medium, or high to control which codes are used in general
DisallowFine tune codes here using “allow” and “disallow” clause with specific codes
AllowFine tune codes here using “allow” and “disallow” clause with specific codes
Codec PriorityCodec priority controls the codec negotiation of an inbound IAX2 call. This option is inherited to all user entity separately which will override the setting in general.

  #

4.4.4 Jitter Buffer Settings #

Jitter Buffer

Instruction of Jitter Buffer

OptionsDefinition
Jitter BufferGlobal default as to whether you want the jitter buffer at all
Force Jitter BufferIn the ideal world, when we bridge VoIP channels we don’t want to jitter buffering on the switch, since the endpoints can each handle this. However, some endpoints may have poor jitter buffers themselves, so this option will force to always jitter buffer, even in this case.
Max Jitter BuffersA maximum size for the jitter buffer
ResyncthresholdWhen the jitter buffer notice a significant change in delay that continue over a few frames, it will resync, assuming that the change in delay was caused by a timestamping mix-up. The threshold for noticing a change in delay is measured as twice the measured jitter plus this resync threshold.
Max Jitter InterpsThe maximum number of interpolation frames the jitter buffer should return in a row. Since some clients do not send CNG/DTX frames to indicate silence, the jitter buffer will assume silence has begun after returning this many interpolations. This prevents interpolating throughout a long silence.
Jitter Target ExtraNumber of milliseconds by which the new jitter buffer will pad its size. The default is 40, so without modification, the new jitter buffer will set its size to the jitter value may help if your network normally has low jitter, but occasionally has spikes.

4.4.5 Misc Settings #

Misc Settings

Instruction of Misc Settings

OptionsDefinition
IAX Thread CountEstablishes the number of iax helper thread to handle I/O
IAX Max Thread CountEstablishes the number of extra dynamic threads that may by spawned to handle I/O
Max Call NumberThe ‘maxcallnumbers’ option limits the amount of call numbers allowed for each individual remote IP address. Once an IP address reaches its call number limit, no more new connections are allowed until the previous ones close. This option can be used in a peer definition as well, but only takes effect for the IP of a dynamic peer after it completes registration.
MaxCallNumbers_NonvalidatedThe ‘maxcallnumbers-nonvalidated’ is used to set the combined number of call numbers that can be allocated for connections where call token validation has been disabled. Unlike the ‘maxcallnumbers’ option, this limit is not separate for each individual IP address. Any connection resulting in a non-call token validated call number being allocated contributes to this limit. For use cases, see the call should be sufficient in most cases.

  #

4.4.6 Quality of Service #

Quality of Service

Instruction of Quality of Service

OptionsDefinition
TosType of service
CosClass of service

4.5 Sip Account Security #

You can configure TLS here. TLS (Transport Layer Security) is a network security protocol used to encrypt and secure data transmission over the internet. It establishes an encrypted channel between two communicating devices (e.g. server and client) to ensure that transmitted data is not intercepted or tampered with.

TLS setting



5. Routing #

Routing Rules

You are allowed to set up new routing rule by , and after setting routing rules, move rules’ order by pulling  up and down, click   button to edit the routing and   to delete it. Finally click the   button to save what you set.
Call Routing Rule:
You can click  button to set up your routing.

Example of Set up Routing Rule

The figure above shows that all the phones in group-aa are transferred to the SIP-1001 terminal.

Definition of Routing Options

OptionsDefinition
Routing NameThe name of this route. Should be used to describe what types of calls this route matches (for example, ‘SIP2CDMA’ or ‘CDAM2SIP’).
Call Comes in FromThe launching point of incoming calls.
Send Call ThroughThe destination to receive the incoming calls.

Description of Advanced Routing Rule

OptionsDefinition
Dial Patterns that will use this RouteA Dial Pattern is a unique set of digits that will select this route and send the call to the designated trunks. If a dialed pattern matches this route, no subsequent routes will be tried. If Time Groups are enabled, subsequent routes will be checked for matches outside of the designated time(s).Rules: X matches any digit from 0-9 Z matches any digit from 1-9 N matches any digit from 2-9 [1237-9] matches any digit in the brackets (example: 1,2,3,7,8,9) . wildcard: matches one or more dialed digits. prepend: Digits to prepend to a successful matchIf the dialed number matches the patterns specified by the subsequent columns, then this will be prepended before sending to the trunks prefix: Prefix to remove on a successful matchThe dialed number is compared to this and the subsequent columns for a match. Upon a match, this prefix is removed from the dialed number before sending it to the trunks. match pattern: The dialed number will be compared against the prefix + this match pattern. Upon a match, the match pattern portion of the dialed number will be sent to the trunks CallerID: If CallerID is supplied, the dialed number will only match the prefix + match pattern if the CallerID has been transmitted matches this.When extensions make outbound calls, the CallerID will be their extension number and NOT their Outbound CID.The above special matching sequences can be used for CallerID matching similar to other number matches.
Set the Caller ID Name toWhat caller ID name would you like to set before sending this call to the endpoint.
Forward NumberWhat destination number will you dial? This is very useful when you have a transfer call.
Custom ContextUser-defined dialing rules
Failover Call Through NumberThe 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.

Time Patterns that will use this Route

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

Failover Call Through Number

You can add one or more “Failover Call Through Numbers”.


5.1 Groups #

Sometimes you want to make a call through one port, but you don’t know if it is available, so you have to check which port is free. That would be troublesome. But with our product, you don’t need to worry about it. You can combine many Port or SIP to groups. Then if you want to make a call, it will find available port automatically.

Routing Group


5.2 Batch Creating rules #

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

Batch Creating rules Group

You can configure the SIM Number, SIP trunk and calling Number for each port.And then, click “save” to batch creating multiple Routing rules.By an attention, the SIP trunk must be configures and the SIM number and calling Number can be emply.

Description of Advanced Routing Rule

OptionsDefinition
Sim NumberWhat destination number will you dial? This is very useful when you have a transfer call.
SIP TrunkInbound and outbound calls through designated SIP trunks
CallerID Make only caller ID to call.

5.3 MNP Settings #

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

MNP Settings


5.4 Route Blacklist #

You can enter numbers here. When these numbers call to your device, device will hang up it.

Routing Blacklist


5.5 Advanced #

You can set dial timeout and call interval here

General

You can edit voice for DISA here

Sounds



6. SMS #

6.1 General #

You can choose enable SMS Process, SMS Local Stored and SMS Status Report or not.

SMS Settings

6.1.1 Sender Options #

You can change sender options here, include resend, times of resend.

Sender Options

Description of Sender Options

OptionsDefinition
Resend Failed MessageThe times that you will attempt to resend your failed message.
Repeat Same MessageThe times that you will resend the same message.

6.1.2 SMS to Email #

This is a tool that makes it available for you to email account to transmit the SMS to other email boxes. The following settings realize that received SMS through a1@fiberme.com transmit to b1@fiberme.com, c1@fiberme.com, and d1@fiberme.com

SMS to Email

You can configure different email in different port

SMS to Email-Destination Email

Types of E-mail Box

E-mail Box TypeSMTP ServerSMTP PortSMTP Security Connectivity
Gmailsmtp.gmail.com587
HotMailsmtp.live.com587
Yahoo!smtp.mail.yahoo.co.in587×
e-mailsmtp.163.com25×

Definition of SMS to E-mail

OptionsDefinition
EnableWhen you choose on, the following options are available, otherwise, unavailable.
Email Address of SenderTo set the email address of an available email account. For example, abc@fiberme.com.
DomainTo set outgoing mail server. e.g. smtp.gmail.com
SMTP PortTo set port number of outgoing mail server. (Default is 25)
SMTP User NameThe 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 PasswordThe password to login your existing email.
TLS EnableWhen you choose Yahoo and 163 free e-mails, this option is not available.
SMTP ServerTo set outgoing mail server. e.g. mail.fiberme.com.
Destination Email Address1The first email address to receive the inbox message.
Destination Email Address2The second email address to receive the inbox message.
Destination Email Address3The third email address to receive the inbox message.
TitleYou can use these parameters to set email title: $PHONENUMBER:SMS sender number. $PORT:SMS from which port. $TIME:SMS received time. $MESSAGE:SMS content.
ContentYou can use these parameters to set email content: $PHONENUMBER:SMS sender number. $PORT:SMS from which port. $TIME:SMS received time. $MESSAGE:SMS content.

  #

6.1.3 SMS Control #

Allowing endpoints to send some specific key words and corresponding password to operate the gateway . In default, this function is disabled.

SMS Control

For example, SMS control password is 123456 which has nothing to do with the login password, you can send “get info 123456” to the module’s phone number to get your gateway’s IP information.

Definition of SMS Control

OptionsDefinition
EnableON(enable), OFF(disable)
PasswordThe password to confirm that SMS makes the gateway rebooted, shut down, restored configuration files and get info on this gateway.
SMS FormatFor 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 cleanswitch 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
SMS outbox Auto cleanswitch on: When the size of the SMS outbox record file reaches the max size, the system will cut a half of the file. New record will be retained.
switch off: SMS record will remain, and the file size will increase gradually. default on, max size = 20 MB

6.1.4 Phone Number Query #

This feature is querying phone number of Internal type.You can set password here. When sending message (get phonenumber password) to device, device will reply your phone number.

Phone Number Query

6.1.5 Auto-reply #

Edit text here, when sending SMS to device, it will reply SMS with text.

Auto-reply

6.1.7 HTTP to SMS #

It support http API for sending SMS . You can call API in your program.

HTTP to SMS

6.1.8 SMS to HTTP #

It support http API for receiving SMS , it can push incoming SMS to your program.

SMS to HTTP Settings


6.2 SMS Sender #

You can choose one or more ports to send SMS to the destination number, different numbers should be separated by symbols: ‘\r’, ‘\n’, space character, semicolon and comma. Then you can see much feedback information.

SMS Sender


6.3 SMS Inbox #

On this page, you are allowed to scan, delete, clean up, and export each port’s received SMS. Also you are allowed to check messages by port, phone number, time order and message keywords.

SMS Inbox


6.4 SMS Outbox #

On this page, you are allowed to scan, delete, clean up, and export each port’s received SMS. Also you are allowed to check messages by port, phone number, time order and message keywords.

SMS Outbox


6.5 SMS Forwarding #

Using this feature, you can forward incoming sms to your mobile. You can click   button to add new routing.
Such as:

SMS Forwarding Rules

SMS received by lte-1.1 and lte-1.2, lte-1.4, will be transfered to phone number 18664565204 through port lte-1.8 or lte-1.10.

For the “ascending” Policy, if you choose 2 or more port members, it will use first available port to transfer sms. For this case, if cdma-1.8 is available, it will always use cdma-1.8 to transfer sms; Otherwise, it will use cdma-1.10 to transfer sms.


  #

6.6 HTTP To USSD #

It support http API for sending USSD . You can call API in your program.

HTTP To USSD


6.7 USSD Result #

It support http API for USSD result , it can push USSD result to your program.

USSD Result


6.8 MMS #

Enter APN here, you can receive MMS.

MMS


6.9 SMPP #

Edit SMPP username and password here. Then you can use send and receive SMS through SMPP

SMPP



7. Network #

7.1 LAN Settings #

There are three types of LAN port IP, Factory, Static and DHCP. Factory is the default type, and it is 172.16.98.1. When you Choose LAN IPv4 type is “Factory”, this page is not editable.
A reserved IP address to access in case your gateway IP is not available. Remember to set a similar network segment with the following address of your local PC.

LAN Settings

Definition of LAN Settings

OptionsDefinition
InterfaceThe name of network interface.
TypeThe 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.
MACPhysical address of your network interface.
AddressThe IP address of your gateway.
NetmaskThe subnet mask of your gateway.
Default GatewayDefault getaway IP address.
Layer 2 QoS 802.1Q/VLAN TagAssigns the VLAN Tag of the Layer 2 QoS packets.Range of 4 to 4095
Default RouteSelect Yes will use current network card DNS and Route

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


7.2 WAN Settings #

There are two types of WAN port IP, Static and DHCP. DHCP is the default type. When you Choose IPv4 type is “Disable” or “DCHP”, this page is not editable.

WAN Settings

Definition of WAN Settings

OptionsDefinition
InterfaceThe name of network interface.
TypeThe 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.
MACPhysical address of your network interface.
AddressThe IP address of your gateway.
NetmskThe subnet mask of your gateway.
Default GatewayDefault getaway IP address.
Layer 2 QoS 802.1Q/VLAN TagAssigns the VLAN Tag of the Layer 2 QoS packets.Range of 4 to 4095
Default RouteSelect Yes will use current network card DNS and route

 7.3 VPN Settings #

FGX4508 series gateways support these VPN.

VPN Settings

Definition of PPTP VPN Settings

OptionsDefinition
VPN TypeNone – close VPN
PPTP VPN – use PPTP VPN
serverThe server’s IP address
AccountServer account
PasswordThe server’s password
Use MPPEWhether to use MPPE
Connection StatusIs it successful to connect to the server

7.4 DDNS Settings #

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

DDNS Settings

Definition of DDNS Settings

OptionsDefinition
DDNSEnable/Disable DDNS(dynamic domain name server)
TypeSet the type of DDNS server.
UsernameYour DDNS account’s login name.
PasswordYour DDNS account’s password.
Your domainThe domain to which your web server will belong.

7.5 Tools #

7.5.1 Ping and Traceroute #

It is used to check network connectivity. Support Ping command on web GUI.

Toolkit

7.5.2 Channel Recording #

You can capture the network packets on the page to facilitate problems.

Capture

 Definition of Channel Recording settings

OptionsDefinition
InterfaceYou can choose eth0 or eth1
Source hostSource host IP
Destination hostDestination host IP
PortWhich port you want to capture
ProtocolWhich protocol you want to capture

 7.6 Security Settings #

7.6.1 Firewall Settings #

Firewall Settings

Definition of Firewall Settings

OptionsDefinition
Firewall EnableIf you want to use White/Black List, and security rules, you must enable this option.
Ping EnableTo disable ping or not. OFF: disable ping. This gateway will not allow to ping.

7.6.2 Settings #

White List Enable: To enable white list or not.
List IP Settings: IPs are separated only by “,” character.

White/Black List Settings

Click “Save” button to save configuration; Click “submit” button to submit and apply configuration.
If “List IP Settings” has no problem, you will see popup window like below. Please read the warning and tips carefully. And Click “Apply” button in 1 minute. If time runs out, this window will close automatically.

Firewall Rules Apply

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

Firewall Rules Apply


7.7 Firewall Rules #

Security Rules

Click “submit” button to submit and apply configuration.
If “List IP Settings” has no problem, you will see popup window like below. Please read the warning and tips carefully. And Click “Apply” button in 1 minute. If time runs out, this window will close automatically.


7.8 SIP Capture #

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

SIP Capture

SIP Capture Settings

OptionsDefinition
InterfaceYou can choose eth0 or eth1
Method-filterYou can choose INVITE, OPTIONS and REGISTER

 7.9 Static Route #

Static Route



8. Advances #

8.1 Asterisk API #

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

Asterisk API

Definition of Asterisk API

OptionsDefinition
PortNetwork port number
Manager NameName of the manager without space
Manager secretPassword 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
PermitIf 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
SystemGeneral information about the system and ability to run system management commands, <br/>such as Shutdown, Restart, and Reload.
CallInformation about channels and ability to set information in a running channel.
LogLogging information. Read-only. (Defined but not yet used.)
VerboseVerbose information. Read-only. (Defined but not yet used.)
CommandPermission to run CLI commands. Write-only.
AgentInformation about queues and agents and ability to add queue members to a queue.
UserPermission to send and receive UserEvent.
ConfigAbility to read and write configuration files.
DTMFReceive DTMF events. Read-only.
ReportingAbility to get information about the system. CDR Output of cdr, manager, if loaded.
CDRCall records. Read-only.
DialplanReceive NewExten and Varset events. Read-only.
OriginatePermission to originate new calls. Write-only.
AllSelect all or deselect all.

Once you set like the above figure, the host 172.16.100.110/255.255.0.0 is allowed to access the gateway API. Please refer to the following figure to access the gateway API by telnet. 172.16.179.1 is the gateway’s IP, and 5038 is its API port.


8.2 Balance #

We offer three ways to query balance: SMS, USSD, Tel.

Note: This feature should be supported by operator firstly.

Balance


8.3 Phone Number #

We offer these ways to query phone number: SMS, USSD, Tel, gateway internal query.

Note: This feature should be supported by operator firstly when you use SMS, USSD, Tel type.

Phone Number


8.4 SNMP #

SNMP is supported. Enable it, then you can get call info.

SNMP


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.

Log Settings

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

CDR Output

definition of Logs

OptionsDefinition
System LogsWhether enable or disable system log.
Auto clean
(System Logs)
switch on : when the size of log file reaches the max size, the system will cut a half of the file. New logs will be retained;
switch off : logs will remain, and the file size will increase gradually. default on, maxsize=1M.
SIP LogsWhether enable or disable SIP log.
Auto clean
(SIP logs)
switch on: when the size of log file reaches the max size, the system will cut a half of the file. New logs will be retained.
switch off: logs will remain, and the file size will increase gradually. default on, maxsize=100KB.
IAX LogsWhether enable or disable IAX log.
Auto clean( IAX logs)switch on: when the size of log file reaches the max size, the system will cut a half of the file. New logs will be retained.
switch off: logs will remain, and the file size will increase gradually. default on, maxsize=100KB.
Call Detail RecordDisplaying Call Detail Records for each channel.
Auto clean (CDR logs)switch on : when the size of log file reaches the max size, the system will cut a half of the file. New logs will be retained.
switch off : logs will remain, and the file size will increase gradually. default on, max size=20MB.
SyslogConfigure it, logs will be sent to your syslog server.


Appendix Feature List #

General Info #

  • LAN:1
  • WAN:1
  • USB Interface:1
  • SIM Cards: hot-swap
  • Temperature: -20~70°C (Storage)     0~40°C (Operation)
  • Operation humidity: 10% ~ 90%non-condensing

VOIP Characters #

  • Support SIP, IAX2 Protocol
  • Add, Modify & Delete SIP/IAX2 Trunk
  • SIP/IAX2 Registration with Domain
  • Combine Different SIP/IAX2 Trunk into Group
  • DTMF Mode: RFC2833/Inband/SIPInfo
  • SIP V2.0 RFC3261 Compliance
  • Multiple SIP/IAX2 Registrations modes:

None (No registration, just IP and Password authenication)
Endpoint registers with this gateway (work as a SIP Sever)
This gateway registers with the endpoint (work as a SIP/IAX2 client)

Network #

  • IPv4,UDP/TCP,DHCP,TELNET,HTTP/HTTPS,TFTP
  • PPTP VPN
  • HTTP/SSH(Optical Telnet)
  • Ping & Traceroute Command on the Web
  • Simple Security Strategy: white list, black list, security rules

System Features #

  • Combine Different SIP/IAX2 Trunk into Group
  • CLID Display & Hide (Need operators’ support )
  • Random call interval
  • Call Duration Limitation
  • Single Call Duration Limitation
  • Real Open API Protocol (based on Asterisk)
  • Support DISA
  • SMSC/SMS/USSD
  • PIN Identification
  • Optional Voice Codec
  • Ports Group Management
  • SMS Bulk Transceiver, Sent to Email and Automatically Resend
  • SMS Coding/Detecting Automatically Identification
  • SMS Remotely Controlling Gateway
  • SMS Forwarding and Quick Reply
  • USSD transceiver
  • Outbound
  • Automatically Reboot
  • Support MMP
  • Support for custom scripts, dialplans

Management #

  • Simple and convenient configuration via Web GUI
  • Support maintenance and configuration by SSH
  • Support configuration files backup and upload
  • Support Chinese and English page
  • Firmware Update by HTTP
  • Support Web and SSH login password modification
  • Restore Factory Settings
  • CDR(More than 200,000 Lines CDRs Storage Locally)
  • System log
  • SIP/IAX2 log
  • TCP and SIP capture
What are your feelings