Version

Release Date

Description

1.0

10/5/2018

Full text

Model

Module

Ports

Network Interface

Band

USB

TF

Console

SWG-2016C

CDMA

16

2

800MHz@CDMA 2000

1

1

1

SWG-2016G

GSM

16

2

850/900/1800/1900MHz@GSM

1

1

1

SWG-2016L

LTE

16

2

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

1

1

1

SWG-2032C

CDMA

32

2

800MHz@CDMA 2000

1

1

1

SWG-2032G

GSM

32

2

850/900/1800/1900MHz@GSM

1

1

1

SWG-2032L

LTE

32

2

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

1

1

1

LED Indicator/Icon/Buttons

Color/ Icon

Staus

Display Icon

Image Added

Module Initiating,Disable

Image Added

No SIM Card

Image Added

Searching for Signal

Image Added

One grid Signal

Image Added

Two grid Signal

Image Added

Three grid Signal

Image Added

four grid Signal

Image Added

fives grid Signal

Image Added

Worst Signal Quality During a Call

Image Added

Medium Signal Quality During a Call

Image Added

Best Signal Quality During a Call

Network Status LED

Green and Flash

Network Connected

Power LED

Always Green

Power on

PER Button

OFF

Power down

ON

Power on

RST Button

–

Press and hold the RST button for 3-5 seconds. The display jumps to the "System Booting" page to restart the system.

Options

Definition

Port

Number of each ports. 

Signal

Display the signal strength of in each channels of gateway. 

BER

Bit Error Rate. 

Carrier

Display the network carrier of current SIM card. 

Registration Status

Indicates the registration status of current module.

PDD

Post Dial Delay (PDD) is experienced by the originating customer as the time from the sending of the final dialed digit to the point at which they hear ring tone or other in-band information.Where the originating network is required to play an announcement before completing the call then this definition of PDD excludes the duration of such announcements. 

ACD

The Average Call Duration (ACD) is calculated by taking the sum of billable seconds (bill sec) of answered calls and dividing it by the number of these answered calls. 

ASR

Answer Seizure Ratio is a measure of network quality. Its calculated by taking the number of successfully answered calls and dividing by the total number of calls attempted. Since busy signals and other rejections by the called number count as call failures, the ASR value can vary depending on user behavior. ModuleStatus Show the status of port, include blank space and "READY". Black space means it is unavailable here and "Ready" means the port is available

Module Status

Display the status of the port. "Ready" means registering and "READY" means port is available

Remain Time

This value is multiplied by to step length is a rest call time.

Options

Definition

System Time

Your gateway system time

Time Zone

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

POSIX TZ String

Posix time zone strings.

NTP Server 1

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

NTP Server 2

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

NTP Server 3

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

Save Data

Save the Modify of the time settings

Sync from NTP

Sync time from NTP server. 

Sync from Client

Sync time from local machine.

Options

Definition

User Name

Define your username and password to manage your gateway
Allowed characters "-_+. < >&0-9a-zA-Z". Length: 1-32 characters.

Password

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

Confirm Password

Please input the same password as 'Password' above.

Login Mode

http and https: You can access gateway via link: http://gatewayIP or https://gatewayIP
https: You can only access gateway via link: https://gatewayIP 

Port

Specify the web server port number.

Options

Definition

Name

The alias of the each port. Input name without space here.
Allowed characters "-_+.<>&0-9a-zA-Z".Length: 1-32 characters. 

Speaker Volume

The speaker volume level, the range is 0-100.
This will adjust the loud speaker volume level by an AT command. 

Microphone Volume

The microphone volume, range is: 0-15.
This will change the microphone gain level by an AT command. 

Dial Prefix

The prefix number of outgoing calls from this channel 

PIN Code

 Personal identification numbers of SIM card. PIN code can be modified to prevent SIM card from being stolen. 

Custom AT commads when start

User custom AT commands when start system, use "

" to split AT command. 

CLIR

 Caller ID restriction, this function is used to hidden caller ID of SIM card number. The gateway will add '#31#' in front of mobile number. This function must support by Operator.

SMS Center Number

Your SMS center number of your local carrier. 

Module IMEI

Only CDMA module does not support modifying IMEI

Options

Definition

Step

Step length value range is 1-999s, step length multiplied by time of single call just said a single call duration time allowed. 

Enable Single Call Duration Limit

Definite maximum call duration for single call. Example: if Time of single call set to 10, the call will be disconnected after talking 10*step seconds.

Enable Call Duration Limitation

This function is to limit the total call duration of channel. The max call duration is between 1 to 999999 minutes. 

Minimum Charging Time

A single call over this time, Module side of the operators began to collect fees, unit for seconds. 

Alarm Threshold

 Define a threshold value of call minutes, while the call minutes less than this value, the gateway will send alarm information to designated phone.

Alarm Description

 Alarm port information description, which will be sent to user mobile phone with alarm information.

Alarm Phone Number

Receiving alarm phone number, user will received alarm message from gateway.

Enable Auto Reset

Automatic restore remaining talk time, that is, get total call minutes of each channel. 

Auto Reset Type

Reset call minutes by date, by week, by month. 

Next Reset Time

  Defined next reset date, system will count start from that date and work as Reset Period setting

Options

Definition

DTMF Normal Twist and Reverse Twist

It is the difference in power between the row and column energies. Normal Twist is where the Column energy is greater than the Row energy. Reverse Twist is where the Row energy is greater.

DTMF Relative Peak Row

The value is the smaller and the detection is easier. If you lost some numbers, you can try to put the value down. The adjustment range is 0.02 at a time. 

DTMF Relative Peak Col

The value is smaller and the detection is easier. If you lost some numbers, you can try to put the value down. The adjustment range is 0.1 at a time. 

DTMF Hits Begin

Sampling matching value. You can choose 2 or 3.

DTMF Misses End

The time interval between the two digits you input. Adjust the speed of input. The smaller value represents the shorter intervals.

Options

Definition

Check Number

 Enter a known number (like your mobile phone) to check what number it is of the SIM card. Click "Execute", then the gateway will dial to the number you already input. It only rings for one time and hangs up at once. Not generating telephone charge during this procedure.

Get USSD

Enter a specific USSD number (For example,*142# to check your SIM card's balance. This USSD number is might be different from different carriers) to get the USSD information. The gateway will try to get by AT commands. 

AT Command

To perform some specific AT commands. This is useful when you have a debug of the modem. e.g. perform [ AT+CSQ ] to check what signal qualify it is. In AT commands, there is no difference between "a" and "A"

Options

Definition

Name

Display name

Username

Register name in your SIP server

Password

Authenticating with the gateway and characters are allowed.

Registration

None --- Not registering;    Server --- When register as this type, it means the gateway acts as a SIP server, and SIP endpoints register to the gateway;   Client --- When register as this type, it means the gateway acts as a client, and the endpoint should be register to a SIP server;

Hostname or IP Address

IP address or hostname of the endpoint or 'dynamic' if the endpoint has a dynamic IP address. This will require registration.

Transport

This sets the possible transport types for outgoing. Order of usage, when the respective transport protocols are enabled, is UDP, TCP, TLS. The first enabled transport type is only used for outbound messages until a Registration takes place. During the peer Registration, the transport type may change to another supported type if the peer requests so.

NAT Traversal

No --- Use Rport if the remote side says to use it. Force Rport on --- Force Rport to always be on. Yes --- Force Rport to always be on and perform comedia RTP handling. Rport if requested and comedia --- Use Rport if the remote side says to use it and perform comedia RTP handling.

Options

Definition

Authentication User

A username to use only for registration.

Register Extension

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

From User

A username to identify the gateway to this endpoint.

From Domain

A domain to identify the gateway to this endpoint.

Remote Secret

A password which is only used if the gateway registers to the remote side.

Port

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

Qualify

Whether or not to check the endpoint's connection status

Qualify Frequency

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

Outbound Proxy

A proxy to which the gateway will send all outbound signalling instead of sending signalling dirrectly to endpoints.

Options

Definition

DTMF Mode

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

Trust Remote-Party-ID

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

Send Remote-Party-ID

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

Remote Party ID Format

How to set the Remote-Party-ID header: from Remote-Party-ID or from P-Asserted-Identity.

Caller ID Presentation

Whether or not to display Caller ID.

Call Limit

Usually used when this sip work as a trunk. To limit number of maximum channels supported by the sip trunk.

Options

Definition

Progress Inband

Whether there is ringing tone.
Never: Indicates that incoming calls are never applicable.
Optional values: yes / no / never. Default: yes

Append user=phone to URI

 Whether or not to Add 'user = phone' to UPIS to include a valid phone number in the URI. 

Add Q.850 Reason Headers

If it is available, Whether or not to add a reason header and use it.

Honor SDP Version

Whether or not to display Caller ID. 

Allow Transfers

Whether or not to globally enable transfers. Choosing 'no' will disable all transfers (unless enabled in peers or users). Default is enabled.

Allow Promiscuous Redirects

Whether or not to allow 302 or REDIR to non-local SIP address. Note that promiscredir when redirects are made to the local system will cause loops since this gateway is incapable of performing a "hairpin" call.

Max Forwards

 Setting for the SIP Max-Forwards header (loop prevention). Send TRYING on REGISTER Send a 100 Trying when the endpoint registers. 

Outbound Proxy

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

Options

Definition

Default T1 Timer

This timer is used primarily in INVITE transactions. The default for Timer T1 is 500ms or the measured run-trip time between the gateway and the device if you have qualify=yes for the device.

Call Setup Timer

 If a provisional response is not received in this amount of time, the call will auto-congest. Defaults to 64 times the default T1 timer. 

Session Timers

Session-Timers feature operates in the following three modes: originate, Request and run session-timers always; accept, run session-timers only when requested by other UA; refuse, do not run session timers in any case.

Minimum Session

 Minimum session refresh interval in seconds. Default is 90secs.

Maximum Session Refresh Interval

Maximum session refresh interval in seconds. Defaults to 1800secs.

Session Refresher

The session refresher, uac or uas. Defaults to uas.

Options

Definition

Name

Display name

Username

Authenication name in your IAX2 server

Password

Authenticating with the gateway and characters are allowed.

Registration

None --- Not registering;    Endpoint registers with this gateway --- When register as this type, it means the gateway acts as a IAX2 server, and IAX2 endpoints register to the gateway;   This gateway registers with the endpoint --- When register as this type, it means the gateway acts as a IAX2 client, and the endpoint should be register to a IAX2 server;

Hostname or
IP Address

IP address or hostname of the endpoint or 'dynamic' if the endpoint has a dynamic IP address. This will require registration.

Auth

There are three authentication methods that are supported: md5, plaintext and rsa. The least secure is "plaintext", which sends passwords cleartext across the net. "md5" uses a challenge/response md5 sum arrangement, but still requires both ends have plain text access to the secret. "rsa" allows unidirectional secret knowledge through public/private keys.If "rsa" authentication is used, "inkeys" is a list of acceptable public keys on the local system that can be used to authenticate the remote peer, separated by the ":" character. "outkey" is a single, private key to use to authenticate to the other side.

Transfer

 This application allows you to transfer calls.

Trunk

"trunk=yes" Purpose: To obtain a better chart of actual bandwidth usage per codec as seen "on-the-wire" when using IAX2 trunking between two Asterisk telephony servers.

Options

Definition

Qualify, Qualify Freq Ok, Qualify Freq Not Ok

The qualify, qualifyfreqok and qualifyfreqnotok settings are used to determine the status availability of an IAX peer. If a peer is consdered to be in a reachable (OK or LAGGED) state, it is queried for availability every "qualifyfreqok" milliseconds. If it is considered to be in an UNREACHABLE state, it is queried for availability every "qualifyfreqnotok" milliseconds.The qualify= setting turns the qualify system on (if the "yes" or xxx options are used) or off (if qualify=no, which is by default). The millisecond value of the qualify= setting specifies the maximum response time of the availability acknowledgement before the peer is considered to be in a "LAGGED" state.

Qualify Smothing

Use an average of the last two PONG result to reduce falsely detected LAGGED host. The default is 'no'.

Port

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

Options

Definition

Encryption

Enable IAX2 encryption. The default is no. 

Force Encryption

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

Options

Definition

Trunk Max Size

 Defaults to 128000 bytes, which supports up to 800; calls of ulaw at 20ms a frame. 

Trunk MTU

With a large amount of traffic on IAX2 trunk, there is a risk of bad voice quality when allowing the Linux system to handle fragmentation of UDP packets. Depending on the side of each payload, allowing the OS to handle fragmentation may not be very efficient. This setting sets the maximum transmission unit for AIX2 UDP trunking. The default is 1240 bytes which means if a trunk's payload is over 1240 bytes for every 20ms it will be broken into multiple 1240 bytes messages. Zero disables this functionality and let's the OS handle fragmentation.

Trunk Frequency

How frequently to send trunk msgs (in ms). This is 20ms by default. 

Trunk Time Stamps

Should we send timestamps for the individual sub_frames within trunk frames? There is a small bandwith use for these (less than 1kbps/call), but they ensure that frame timestamps get sent end-to-end properly. If both ends of all your trunks go directly to TDM, _and_your trunkfreq equals the frame length for your codecs, you can probably suppress these. The receiver must also need to have it enabled.

Min. RegExpire

Minimum amounts of time that IAX2 peers can request as a registration interval (in seconds).

Max. RegExpire

Maximum amounts of time that IAX2 peers can request as a registration expiration interval(in seconds). 

Options

Definition

Name

Display name

Username

Register name in your SIP server

Password

Authenticating with the gateway and characters are allowed.

Registration

None --- Not registering;    Server --- When register as this type, it means the gateway acts as a SIP server, and SIP endpoints register to the gateway;   Client --- When register as this type, it means the gateway acts as a client, and the endpoint should be register to a SIP server;

Hostname or IP Address

IP address or hostname of the endpoint or 'dynamic' if the endpoint has a dynamic IP address. This will require registration.

AutoPassword

Tick - Automatically increments based on the password entered in the first lineDo not check - All SIP endpoints have the same password as the first one.

Options

Definition

UDP Bind Port

UDP Bind Port

Enable TCP

Enable server for incoming TCP connection (default is no).

TCP Bind Port

Choose a port on which to listen for TCP traffic.

TCP Authentication Timeout

The maximum number of seconds a client has to authenticate. If the client does not authenticate before this timeout expires, the client will be disconnected.(default value is: 30 seconds).

TCP Authentication Limit

The maximum number of unauthenticated sessions that will be allowed to connect at any given time (default is: 50).

Enable Hostname Lookup

Enable DNS SRV lookups on outbound calls Note: the gateway only uses the first host in SRV records Disabling DNS SRV lookups disables the ability to place SIP calls based on domain names to some other SIP users on the Internet specifying a port in a SIP peer definition or when dialing outbound calls with suppress SRV lookups for that peer or call.

Enable Internal SIP Call

Whether enable the internal SIP calls or not when you select the registration option "Endpoint registers with this gateway".

Internal SIP Call Prefix

Specify a prefix before routing the internal calls.

Options

Definition

Local Network

Format:192.168.0.0/255.255.0.0 or 172.16.0.0./12. A list of IP address or IP ranges which are located inside a NATed network. This gateway will replace the internal IP address in SIP and SDP messages with the external IP address when a NAT exists between the gateway and other endpoints.

 Local Network List

Local IP address list that you added.

Subscribe Network Change Event

Through the use of the test_stun_monitor module, the gateway has the ability to detect when the perceived external network address has changed. When the stun_monitor is installed and configured, chan_sip will renew all outbound registrations when the monitor detects any sort of network change has occurred. By default this option is enabled, but only takes effect once res_stun_monitor is configured. If res_stun_monitor is enabled and you wish to not generate all outbound registrations on a network change, use the option below to disable this feature.

Match External Address Locally

Only substitute the externaddr or externhost setting if it matches.

Dynamic Exclude Static

Disallow all dynamic hosts from registering as any IP address used for statically defined hosts. This helps avoid the configuration error of allowing your users to register at the same address as a SIP provider.

Externally Mapped TCP Port

The externally mapped TCP port, when the gateway is behind a static NAT or PAT.

External Hostname

The external hostname (and optional TCP port) of the NAT. 

Hostname Refresh Interval

How often to perform a hostname lookup. This can be useful when your NAT device lets you choose the port mapping, but the IP address is dynamic. Beware, you might suffer from service disruption when the name server resolution fails.

Options

Definition

Start of RTP Port Range

 Start of range of port numbers to be used for RTP

End of RTP port Range

End of port numbers to be used for RTP

RTPTimeout

RTP Timeout retransmission time

Options

Definition

Strict RFC Interpretation

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

Send Compact Headers

Send compact SIP headers 

SDP Owner

Allows you to change the username filed in the SDP owner string. This filed MUST NOT contain spaces. 

Disallowed SIP Methods

The external hostname (and optional TCP port) of the NAT. 

Shrink Caller ID

The shrinkcallerid function removes '(', ' ', ')', non-trailing '.', and '-' not in square brackets. For example, the caller id value 555.5555 becomes 5555555 when this option is enabled. Disabling this option results in no modification of the caller id value, which is necessary when the caller id represents something that must be preserved. By default this option is on.

Maximum Registration Expiry

Maximum allowed time of incoming registrations and subscriptions (seconds).

Minimum Registration Expiry

Minimum length of registrations/subscriptions (default 60).

Default Registration Expiry

Default length of incoming/outgoing registration.

Registration Timeout

How often, in seconds, to retry registration calls. Default 20 seconds.

Number of Registration

Attempts Enter '0' for unlimited Number of registration attempts before we give up. 0 = continue forever, hammering the other server until it accepts the registration. Default is 0 tries, continue forever.

Options

Definition

Match Auth Username

If available, match user entry using the 'username' field from the authentication line instead of the 'from' field.

Realm

Realm for digest authentication. Realms MUST be globally unique according to RFC 3261. Set this to your host name or domain name.

Use Domain as Realm

Use the domain from the SIP Domains setting as the realm. In this case, the realm will be based on the request 'to' or 'from' header and should match one of the domain. Otherwise, the configured 'realm' value will be used.

Always Auth Reject

When an incoming INVITE or REGISTER is to be rejected, for any reason, always reject with an identical response equivalent to valid username and invalid password/hash instead of letting the requester know whether there was a matching user or peer for their request. This reduces the ability of an attacker to scan for valid SIP usernames. This option is set to 'yes' by default.

Authenticate Options Requests

Enabling this option will authenticate OPTIONS requests just like INVITE requests are. By default this option is disabled. 

Allow Guest Calling

Allow or reject guest calls (default is yes, to allow). If your gateway is connected to the Internet and you allow guest calls, you want to check which services you offer everyone out there, by enabling them in the default context.

Options

Definition

Premature Media

Some ISDN links send empty media frames before the call is in ringing or progress state. The SIP channel will then send 183 indicating early media which will be empty - thus users get no ring signal. Setting this to "yes" will stop any media before we have call progress (meaning the SIP channel will not send 183 Session Progress for early media). Default is 'yes'. Also make sure that the SIP peer is configured with progressinband=never. In order for 'noanswer' applications 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

Bind Port

Bind port and bindaddr may be specified

Enable IAXCompat

More than once to bind to multiple addresses, but the first will be the default.

Enable Nochecksums

Set iaxcompat to yes if you plan to use layered switches or some other scenario which may cause some delay when doing a lookup in the dialplan. It incurs a small performance hit to enable it. This option cause Asterisk to spawn a separate thread when it receives an IAX DPREQ (Dialplan Request) instead of blocking while it waits for a response.

Enable Delay Reject

Disable UDP checksums (if no checksums is set, then no checksums will be calculated/checked on system supporting the feature)

ADSI

ADSI (Analog Display Services Interface) can be enable if you have (or may have) ADSI compatible CPE equipment.

SRV Loopup

Whether or not to perform an SRV lookup on outbound calls

AMA Flags

You may specify a global default AMA flag for iaxtel calls. These flags are used in the generation of call detail records.

autokill

If we don't get ACK to our NEW within 2000ms,and autokill is set to yes, then we cancel the whole thing(that's enough time for one retransmission only ).This is used to keep things from stalling for a long time for a host that is not available for bad connections.

Language

You may specify a global default language for users. This can be specified also on a per-user basis. If omitted, will fallback to English(en)

Account Code

You may specify a default account for Call Detail Records (CDRs) in addition specifying on a per-user basis.

Options

Definition

Mohsuggest

The 'Mohsuggest' option specifies which music on hold class to suggest to the peer channel when this channel place the peer on hold. It may be specified globally or on a per-user or per-peer basis.

Mohinterpret

You may specify a global default language for users. This can be specified also on a per-user basis. If omitted, will fall back to English(en)

Options

Definition

Band Width

Specify bandwith of low, medium, or high to control which codes are used in general

Disallow

Fine tune codes here using "allow" and "disallow" clause with specific codes

Allow

Fine tune codes here using "allow" and "disallow" clause with specific codes

Codec Priority

Codec priority controls the codec negotiation of an inbound IAX2 call. This option is inherited to all user entity separately which will override the setting in general.

Options

Definition

Jitter Buffer

Global default as to whether you want the jitter buffer at all

Force Jitter Buffer

In the ideal world, when we bridge VoIP channels we don't want to jitter buffering on the switch, since the endpoints can each handle this. However, some endpoints may have poor jitter buffers themselves, so this option will force to always jitter buffer, even in this case.

Max Jitter Buffers

A maximum size for the jitter buffer

Resyncthreshold

When the jitter buffer notice a significant change in delay that continue over a few frames, it will resync, assuming that the change in delay was caused by a timestamping mix-up. The threshold for noticing a change in delay is measured as twice the measured jitter plus this resync threshold.

Max Jitter Interps

The maximum number of interpolation frames the jitter buffer should return in a row. Since some clients do not send CNG/DTX frames to indicate silence, the jitter buffer will assume silence has begun after returning this many interpolations. This prevents interpolating throughout a long silence.

Jitter Target Extra

Number of milliseconds by which the new jitter buffer will pad its size. The default is 40, so without modification, the new jitter buffer will set its size to the jitter value may help if your network normally has low jitter, but occasionally has spikes.

Options

Definition

IAX Thread Count

Establishes the number of iax helper thread to handle I/O

IAX Max Thread Count

Establishes the number of extra dynamic threads that may by spawned to handle I/O

Max Call Number

The 'maxcallnumbers' option limits the amount of call numbers allowed for each individual remote IP address. Once an IP address reaches its call number limit, no more new connections are allowed until the previous ones close. This option can be used in a peer definition as well, but only takes effect for the IP of a dynamic peer after it completes registration.

MaxCallNumbers_Nonvalidated

The 'maxcallnumbers-nonvalidated' is used to set the combined number of call numbers that can be allocated for connections where call token validation has been disabled. 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.

Options

Definition

Tos

Type of service

Cos

Class of service

Options

Definition

Routing Name

The name of this route. Should be used to describe what types of calls this route matches (for example, 'SIP2CDMA' or 'CDAM2SIP').

Call Comes in From

The launching point of incoming calls.

Send Call Through

The destination to receive the incoming calls.

Options

Definition

Dial Patterns that will use this Route

A Dial Pattern is a unique set of digits that will select this route and send the call to the designated trunks. If a dialed pattern matches this route, no subsequent routes will be tried. If Time Groups are enabled, subsequent routes will be checked for matches outside of the designated time(s).Rules: X matches any digit from 0-9 Z matches any digit from 1-9 N matches any digit from 2-9 [1237-9] matches any digit in the brackets (example: 1,2,3,7,8,9) . wildcard: matches one or more dialed digits. prepend: Digits to prepend to a successful 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 to

What caller ID name would you like to set before sending this call to the endpoint.

Forward Number

What destination number will you dial? This is very useful when you have a transfer call.

Custom Context

User-defined dialing rules

Failover Call Through Number

The gateway will attempt to send the call out each of these in the order you specify. You can create various time routes and use these time conditions to limit some specific calls.

Options

Definition

Forward Number

What destination number will you dial? This is very useful when you have a transfer call.

SIP Trunk

Inbound and outbound calls through designated SIP trunks

Set the Caller ID Name to

What caller ID name would you like to set before sending this call to the endpoint.

Options

Definition

Resend Failed Message

The times that you will attempt to resend your failed message. 

Repeat Same Message

The times that you will resend the same message. 

E-mail Box Type

SMTP Server

SMTP Port

SMTP Security Connectivity

Gmail

smtp.gmail.com

587

√

HotMail

smtp.live.com

587

√

Yahoo!

smtp.mail.yahoo.co.in

587

×

e-mail

smtp.163.com

25

×

Options

Definition

Enable

When you choose on, the following options are available, otherwise, unavailable.

Email Address of Sender

To set the email address of an available email account. For example, openvpnvoip@gmail.com.

Domain

To set outgoing mail server. e.g. smtp.gmail.com

SMTP Port

To set port number of outgoing mail server. (Default is 25)

SMTP User Name

The login name of your existing email account. This option might be different from your email address. Some email client doesn't need the email postfix

SMTP Password

The password to login your existing email.

TLS Enable

When you choose Yahoo and 163 free e-mails, this option is not available.

SMTP Server

To set outgoing mail server. e.g. mail.openvox.cn.

Destination Email Address1

The first email address to receive the inbox message.

Destination Email Address2

The second email address to receive the inbox message.

Destination Email Address3

The third email address to receive the inbox message.

Options

Definition

Enable

ON(enable), OFF(disable)

Password

The password to confirm that SMS makes the gateway rebooted, shut down, restored configuration files and get info on this gateway.

SMS Format

For example, the message formats:
reboot system PASSWORD: To reboot your whole gateway.
The PASSWORD is referring to the PASSWORD you set up from option "PASSWORD" above.
Reboot asterisk PASSWORD: To restart your gateway core.
Restore configs PASSWORD: To reset the configuration files back to the default factory settings.
Get info PASSWORD: To get your gateway IP address

SMS inbox Auto clean

switch on: When the size of the SMS inbox record file reaches the max size, the system will cut a half of the file. New record will be retained.
switch off: SMS record will remain, and the file size will increase gradually. default on, max size = 20 MB

Options

Definition

Interface

The name of network interface.

Type

The method to get IP. 
Factory: Getting IP address by Slot Number
(System information to check slot number). 
Static: manually set up your gateway IP. 
DHCP: automatically get IP from your local LAN.

MAC

Physical address of your network interface.

Address

The IP address of your gateway.

Netmsk

The subnet mask of your gateway.

Default Gateway

Default getaway IP address.

Options

Definition

Interface

The name of network interface.

Type

The method to get IP. 
Factory: Getting IP address by Slot Number
(System information to check slot number). 
Static: manually set up your gateway IP. 
DHCP: automatically get IP from your local LAN.

MAC

Physical address of your network interface.

Address

The IP address of your gateway.

Netmsk

The subnet mask of your gateway.

Default Gateway

Default getaway IP address.

Options

Definition

VPN Type

None – close VPN
PPTP VPN – use PPTP VPN

server

The server's IP address

Account

Server account

Password

The server's password

Use MPPE

Whether to use MPPE

Connection Status

Is it successful to connect to the server

Options

Definition

DDNS

Enable/Disable DDNS(dynamic domain name server) 

Type

Set the type of DDNS server. 

Username

Your DDNS account's login name. 

Password

Your DDNS account's password. 

Your domain

The domain to which your web server will belong.

Options

Definition

Inferface

You can choose eth0 or eth1 

Source host

Source host IP

Destination host

Destination host IP

Port

Which port you want to capture?

Protocol

Which protocol you want to capture?