Version

Release Date

Description

1.0

25/8/2021

First Version

Weight（with packaging）

2.86kg（16 ports）

2.98kg（24 ports）

3.1kg（32 ports）

Size（bare）

440*45*220mm

Size（with packaging）

510*90*310mm

Temperature

-20~70°C (Storage)

0~45°C (Operation)

Operation humidity

10%~90% non-condensing

Power source

90-264VAC

Maximum power

20W（16 ports）

28W（24 ports）

32W（32 ports）

Pin No.

Port No.

Pin No.

Port No.

1

1

1

25

26

26

2

2

2

26

27

27

3

3

3

27

28

28

4

4

4

28

29

29

5

5

5

29

30

30

6

6

6

30

31

31

7

7

7

31

32

32

8

8

8

32

33

33

9

9

9

NC

34

34

10

10

10

NC

35

35

11

11

11

NC

36

36

12

12

12

NC

37

37

13

13

13

NC

38

38

14

14

14

NC

39

39

15

15

15

NC

40

40

16

16

16

NC

41

41

17

17

17

NC

42

42

18

18

18

NC

43

43

19

19

19

NC

44

44

20

20

20

NC

45

45

21

21

21

NC

46

46

22

22

22

NC

47

47

23

23

23

NC

48

48

24

24

24

NC

49

49

25

NC

25

NC

50

50

Options

Definition

System Time

Your gateway system time.

Time Zone

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

POSIX TZ String

Posix time zone strings.

NTP Server 1

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

NTP Server 2

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

NTP Server 3

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

Auto-Sync from NTP

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.

Options

Definition

User Name

Define your username and password to manage your gateway, without space here. Allowed characters

"-_+. < >&0-9a-zA-Z". Length: 1-32 characters.

Password

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

Confirm Password

Please input the same password as 'Password' above.

Login Mode

Select the mode of login.

HTTP Port

Specify the web server port number.

HTTPS Port

Specify the web server port number.

Port

SSH login port number.

Options

Definition

System Reboot

This option will restart your gateway and cut off all current sessions.

Asterisk Reboot

This option will restart Asterisk and cut off all current sessions.

Options

Definition

Enable

ON(enabled),OFF(disabled)

Time Out

Set the timeout, in milliseconds (ms).

Notice: You can only enter numbers.

Number

Pickup number

Options

Definition

Signal hold

The duration of the signal tone generated by the corresponding channel. (in milliseconds).

Dial timeout

The specified timeout period (seconds) for dialing a specific device.

Echo algorithm type

Default G168

Signal interval

Default 100 (milliseconds)

Echo cancellation signal length

Software echo cancellation signal length

FXS signaling

Default Loop start, busy tone is generated, Kewlstart, power is off, no busy tone is generated

Options

Definition

model

Set transfer mode

Baud rate

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.

Options

Definition

Country

Set the signal tone standard of the country where the gateway is located.

Ringing rhythm

List of continuous ringing.

Dial tone

Set the off-hook dial tone.

Ringback tone

Set the prompt tone to the caller when ringing.

Busy tone

Set the prompt tone when busy.

Call waiting tone

Set the background prompt tone to play when entering the call waiting.

Congestion tone

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.

Recording tone

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).

Options

Definition

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.

Hook/flash

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.

Options

Definition

Off-hook anti-shake

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.

Options

Definition

Encoding

Global encoding settings: mulaw, alaw

Impedance

Impedance setting

Options

Definition

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

Options

Definition

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

Options

Definition

Name

A name which is able to read. And it’s only used for user’s reference.

Username

Username for authentication between the endpoint and the gateway.

Password

The password for authentication between the endpoint and the gateway, allowing letters.

Registration

None---Anonymous registration;

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

Transmission method

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.

NAT Traversal

Issues related to NAT addresses when incoming SIP or media sessions.

VOS encryption

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

Options

Definition

Authentication User

A username to use only for registration.

Register Extension

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

Registered user name

The registered username，is the user in ”register  => user[:secret[:authuser]]@host[:port][/extension]“

User source

A username to identify the gateway to this endpoint.

From Domain

A domain to identify the gateway to this endpoint.

Port

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

Quality

To check the endpoint's connection status whether or not.

Qualify Frequency

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

Outbound Proxy

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

Customized registration switch

After opening, customers can customize the registration form by themselves.

Enable Outboundproxy to replace Host

Options

Definition

DTMF Mode

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

Call Limit

Set a call limit, the maximum number of calls that can be allowed at the same time.

Trust Remote-Party-ID

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

Send Remote-Party-ID

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

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.

Options

Definition

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.

Allow Transfers

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.

Maximum Forward

Setting for the SIP Max-Forwards header (loop prevention).

Send TRYING on REGISTER

Send a 100 Trying when the endpoint registers.

Options

Definition

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.

Session Timers

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

Session refresher, user agent client or user agent server. The default is the user agent server.

Options

Definition

Media Settings

Select codec from the drop down list. Different encoding priorities choose different encoding methods.

Options

Definition

UDP Bind Port

Choose a port on which to listen for UDP traffic.

Enable TCP

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.

Options

Definition

Local Network

Format:192.168.0.0/255.255.0.0 or 172.16.0.0./12.

A list of IP address or IP ranges which are located inside a 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.

Examples:

External IP Address = 12.34.56.78

External IP Address = 12.34.56.78:9900

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”.

Options

Definition

Start

Turn on function

Server Port

Default port 3478

Refresh Request Interval

 Time interval in seconds, default 30 seconds

Server IP Address/Domain Name

Server address or domain name

Options

Definition

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.

RTP Timeout

 RTP Timeout

Options

Definition

Strict RFC Interpretation

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

Send Compact Headers

Send compact SIP headers

SDP Owner

Allows you to change the domain of the SDP username. £¨o=£©
This filed MUST NOT contain spaces.

Disallowed SIP Methods

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.

Registration Timeout

How long will it take to re-register the extension. (Default 20 seconds)

Number of registration attempts

Number of registration attempts before giving up.

Options

Definition

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.

Realm

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”.

Premature Media

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

Options

Definition

TLS Enable

Enable or disable DTLS-SRTP support.

TLS Verify Server

Enable or disable TLS verify server(default is no).

Port

Specify the port for remote connection.

TLS Client Method

Values include tlsv1, sslv3, sslv2, Specify protocol for outbound client connections (default is sslv2)

Options

Definition

Routing Name

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.

Call Delivery

The destination to receive the incoming calls.

DISA Timeout

The specific setting time of DISA timeout.

Maximum Number of Digits In Password

Set the maximum number of password digits

Password

Set a password within the specified range

Options

Definition

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.

Forward Number

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.

Options

Definition

Group description

The name of the route, used to describe the type of this call route, for example, sip1 TO port1 or port1 TO sip2.

Options

Definition

Network Type

The name of network interface.

Type

The method to get IP.

Static: manually set up your gateway IP.

DHCP: dynamically obtain the gateway IP address.

Physical address

The physical address of the network interface.

Address

The IP address of your gateway.

Netmask

The subnet mask of your gateway.

Default 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

Enable or disable the reserved IP address switch.

ON(enabled), OFF(disabled)

Reserved Address

The reserved IP address for this gateway.

Reserved Netmask

The subnet mask of the reserved IP address.

Options

Definition

DDNS

Enable/Disable DDNS(dynamic domain name server)

Type

Set the type of DDNS server.

Username

Your DDNS account’s login name.

Password

Your DDNS account’s password.

Domain

The domain to which your web server will belong.

Options

Definition

Interface

The name of network interface.

Source Host Address

Specify the source address of the data you want to get

Destination Host

Specify the destination address you want to get data from

Port

Specify the port where you want to get data

Channel

Specify the channel number you want to get data

Tcpdump Option Parameter

The tool of tcpdump capture network data by parameter option specified.

Options

Definition

Port

Network port number

Administrator Name

Name of the manager cannot contain spaces.

Administrator Password

Password for the administrator.

Characters: Allowed characters “-_+.<>&0-9a-zA-Z”.

Length:4-32 characters.

Deny

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

Permit

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

System log

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

Call

Channel information and setting information of the channel in use.

Log

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

Details

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

Command

CLI commands allowed to run. (Read-only)

Agent

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

User

Permission to send and receive UserEvent.

Config

Ability to read and write configuration files.

DTMF

Receive DTMF events. (Read-only)

Reporting

Ability to get information about the system.

CDR

Output of “cdr, manager”, if loaded. (Read-only)

Dial Plan

Receive “NewExten” and “Varset” events. (Read-only)

Originate

Permission to originate new calls. (Read-only)

All

Select all or deselect all.

Options

Definition

System Logs

Whether to open the 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, max size=1MB.

Verbose

Asterisk console verbose message switch.

Notice

Asterisk console notice message switch.

Warning

Asterisk console warning message switch.

Debug

Asterisk console debug message switch.

Error

Asterisk console error message switch.

DTMF

Asterisk console DTMF info switch.

Auto clean:

(asterisk logs)

switch on :

when the size of log file reaches the max size, the system will cut a half of the file. New logs will be retained.

switch off :

logs will remain, and the file size will increase gradually.

default on, max size=100KB.

SIP Logs:

Whether enable or disable SIP log.

Auto clean:

(SIP logs)

switch on :

when the size of log file reaches the max size,the system will cut a half of the file. New logs will be retained.

switch off :

logs will remain, and the file size will increase gradually.

default on, default size=100KB.