Analog Gateway User Manual


1. Overview

What is VS-GW1600-40S?

OpenVox VoxStack Series Analog Gateway is an open source asterisk-based Analog VoIP Gateway solution for SMBs and SOHOs. With friendly GUI and unique modular design, users may easily setup their customized Gateway. Also secondary development can be completed through AMI (Asterisk Management Interface).

There are three models with VoxStack series Analog Gateway, the VS-GW1202-8S, VS-GW1202-16S and VS-GW1600-40S. There are 8 ports in VS-GW1202-8S. The Modular Design Analog Gateways are ranging from 8 up to 40 ports, developed for interconnecting the PSTN networks with a wide selection of codecs and signaling protocol, including G.711A, G.711U, G.729, G.722, G.723, ILBC and GSM to quickly reduce communication expenses and maximize cost-savings. With the unique design of the VoxStack gateway, it can support hot-swap. Users can simply add or remove the modules for hardware expansion or exchange.

The VoxStack gateway designs with 2 LAN switch boards to provide stack ability on the hardware upgrade. You can choose either of them.

The Analog gateway will be 100% compatible with Asterisk, Elastix, trixbox, 3CX, FreeSWITCH SIP server and VOS VoIP operating platform.



Sample Application

Figure 1-2-1 Topological Graph

Product Appearance

The picture below is appearance of Analog Series Gateway.


 Figure 1-3-1 Product Appearance


     VS-GW1202-8S                                                                VS-GW1600-40S


Main Features

  • Modular and VoxStack design
  • Based on AsteriskR
  • Editable AsteriskRconfigurationfile
  •  Support T.38 fax relay and T.30 fax transparent, can continually fax multiple page
  •  Echo cancellation and Static jitter buffer
    • Wide selection of codecs and signaling protocol
    • DTMF relay
    • Ring cadence and frequency setting
    • MWI(Message waiting indicator)
    • DHCP , DNS/DDNS, NAT Network
    • VAG and CNG
    • All hot-swap
    • Stable performance, flexible dialing, friendly GUI
    • Two-year time warranty

Physical Information

Table 1-5-1 Description of Physical Information







-20~70°C (Storage)

0~40°C (Operation)

Operation humidity

10%~90% non-condensing

Power source

12V DC/4A

Max power


LAN port





Default IP:

Username: admin

Password: admin


Please enter the default IP in your browser to scan and configure the module you want. Now we offer you two RJ45 Network ports to access to your gateway on the board, ETH1 and ETH2. You can choose either of them and they are the same.

Figure 1-6-1 LOGIN Interface


2. System


On the “Status” page, you will see Port/SIP/Network information and status.

Figure 2-1-1 System Status


Table 2-2-1 Description of Time Settings




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 time zone strings.

NTP Server 1

Time server domain or hostname. For example, [].

NTP Server 2

The first reserved NTP server. For example, [].

NTP Server 3

The second reserved NTP server. For example, [].

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.



For example, you can configure like this:

                  Figure 2-2-1 Time Settings

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

Login Settings

Your gateway doesn't have administration role. All you can do here is to reset what new username and password to manage your gateway. And it has all privileges to operate your gateway. You can modify both your “Web Login Settings” and “SSH Login Settings”. If you have changed these settings, you don’t need to log out, just rewriting your new user name and password will be OK.


                    Table 2-3-1 Description of Login Settings




User Name

Define your username and password to manage your gateway,

without space here. Allowed characters

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


Allowed characters "-_+. < >&0-9a-zA-Z".

Length: 4-32 characters.

Confirm Password

Please input the same password as 'Password' above.



                      Figure 2-3-1 Login Settings

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

General, Tools and Information

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”, those will be ok.

                   Figure 2-4-1 Language Settings

Scheduled Reboot

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

                   Figure 2-4-2 Reboot Types


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

Reboot Tools

On the “Tools” pages, there are reboot, update, upload, backup and restore toolkits.

You can choose system reboot and Asterisk reboot separately.

                          Figure 2-4-3 Reboot Prompt

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

            Table 2-4-1 Instruction of reboots




System Reboot 

This will turn off your gateway and then turn it back on. This

will drop all current calls.

Asterisk Reboot

This will restart Asterisk and drop all current calls.


We offer two kinds of update types for you, you can choose System Update or System Online Update. System Online Update is an easier way to update your system.

                    Figure 2-4-4 Update Firmware


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

               Figure 2-4-5 Upload and Backup

Sometimes there is something wrong with your gateway that you don’t know how to solve it,

mostly you will select factory reset. Then you just need to press a button, your gateway will be reset to the factory status.

              Figure 2-4-6 Factory Reset


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

                    Figure 2-4-7 System Information

3. Analog

You can see much information about your ports on this page.

Channel Settings

Figure 3-1-1 Channel System


On this page, you can see every port status, and click action  button to configure the port.

Figure 3-1-2 Port Configure


Dial Matching Table

Dialing rules is used to effectively judge whether the received number sequence is complete, in order to timely end receiving number and send out number

The correct use of dial-up rules, helps to shorten the turn-on time of phone call

Figure 3-2-1 Port Configure


Global Settings

               Figure 3-3-1 General Configuration

            Table 3-3-1 Instruction of General




Tone duration

How long generated tones (DTMF and MF) will be played on the channel.

(in milliseconds)

Dial timeout

Specifies the number of seconds we attempt to dial the specified devices.


Set the global encoding : mulaw, alaw.


Configuration for impedance.

Echo cancel tap length

Hardware echo canceler tap length.


Turn on/off VAD/CNG.


Turn on/off Flash/wink.

Max flash time

Max flash time.(in milliseconds).

“#”as Ending Dial Key

Turn on/off Ending Dial Key.


          Figure 3-3-2 Caller ID 



Table 3-3-2 Instruction of Caller ID

Options Definition
The pattern of sending CID

Some countries(UK) have ring tones with different ring tones(ring-ring),

which means the caller ID needs to be set later on, and not just after the

first ring, as per the default(1) 

Waiting time before sending CID

How long we will waiting before sending the CID on the channel.

(in milliseconds).

Sending polarity reversal(DTMF Only)

Send polarity reversal before sending the CID on the channel.

Start code(DTMF Only)Stop code.
Stop code(DTMF Only)Stop code.



Figure 3-3-3 Hardware Gain


Table 3-3-3 Instruction of Hardware gain




FXS Rx gain

Set the FXS port Rx gain. Range: -35, 0 or 35.

FXS Tx gain

Set the FXS port Tx gain. Range: -35, 0 or 35.



Figure 3-3-4 Fax Configuration


Table 3-3-4 Definition of Fax


ModeSet the transmission mode.
RateSet the rate of sending and receiving.
EcmEnable/disable T.30 ECM (error correction mode) by default.




Figure 3-3-5 Country Configuration



Table 3-3-5 Definition of Country


Options Definition
CountryConfiguration for location specific tone indications.
Ring cadenceList of durations the physical bell rings.
Dial toneSet of tones to be played when one picks up the hook.
Ring toneSet of tones to be played when the receiving end is ringing.
Busy toneSet of tones played when the receiving end is busy.
Call waiting toneSet of tones played when there is a call waiting in the background.
Congestion toneSet of tones played when there is some congestion.
Dial recall toneMany phone systems play a recall dial tone after hook flash.
Record toneSet of tones played when call recording is in progress.
Info toneSet of tones played with special information messages (e.g., number is out of service.)
Stutter tone 




4. SIP

SIP Endpoints

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


Figure 4-1-1 SIP Status


You can click button to add a new SIP endpoint, and if you want to modify existed endpoints, you can click  button.

Main Endpoint Settings

There are 3 kinds of registration types for choose. You can choose “Anonymous, Endpoint registers with this gateway or This gateway registers with the endpoint”.


You can configure as follows:

If you set up a SIP endpoint by registration “None” to a server, then you can’t register other SIP endpoints to this server. (If you add other SIP endpoints, this will cause Out-band Routes and Trunks confused.)


Figure 4-1-2 Anonymous Registration


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


Figure 4-1-3 Register to Gateway


Also you can choose registration by “This gateway registers with the endpoint”, it’s the same with


“None”, except name and password.


Figure 4-1-4 Register to Server


Table 4-1-1 Definition of SIP Options


Options Definition
NameA name which is able to read by human. And it’s only used for user’s reference.
UsernameUser Name the endpoint will use to authenticate with the gateway.

Password the endpoint will use to authenticate with the gateway. Allowed



None---Not registering;

Endpoint registers with this gateway---When register as this type, it means

the GSM gateway acts as a SIP server, and SIP endpoints register to the gateway;

This gateway registers with the endpoint---When register as this type, it means

the GSM 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.


This sets the possible transport types for outgoing. Order of usage, when the

respective transport protocols are enabled, is UDP, TCP, TLS. The first enabled

transport type is only used for outbound messages until a Registration takes place.

During the peer Registration the transport type may change to another supported

type if the peer requests so.

NAT Traversal

Addresses NAT-related issues in incoming SIP or media sessions.

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


Rport if requested and comedia---Use Rport if the remote side

says to use it and perform comedia RTP handling.




Advanced: Registration Options

Table 4-1-2 Definition of Registration Options




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.


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


Whether or not to check the endpoint's connection status.

Qualify Frequency

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



Call Settings

Table 4-1-3 Definition of Call Options





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

Call Limit

Setting a call-limit will cause calls above the limit not to be accepted.

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


Caller ID Presentation

Whether or not to display Caller ID.



Advanced: Signaling Settings

Table 4-1-4 Definition of Signaling Options


Options Definition
Progress InbandSet 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).
Allow Overlap DialingAllow Overlap Dialing: Whether or not to allow overlap dialing. Disabled by default.
Append user=phone to URIWhether or not to add ‘; user=phone’ to URIs that contain a valid phone number.
Add Q.850 Reason HeadersWhether or not to add Reason header and to use it if it is available.
Honor SDP Version

By default, the gateway will honor the session version number in SDP packets

and will only modify the SDP session if the version number change. Turn this

option off to force the gateway to ignore the SDP session version number and

treat all SDP data as new data. This is required for devices that send non-standard

SDP packets (observed with Microsoft OCS). By default this option is on.

Allow Transfers

Whether or not to globally enable transfers. Choosing 'no' will disable all transfers

(unless enabled in peers or users). Default is enabled.

Allow Promiscuous Redirects

Whether or not to allow 302 or REDIR to non-local SIP address. 
Note that promiscredir when redirects are made to the local system will cause

loops since this gateway is incapable of performing a "hairpin" call.

Max ForwardsSetting for the SIP Max-Forwards header (loop prevention).
Send TRYING on REGISTERSend 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.





Advanced: Timer Settings

Table 4-1-5 Definition of Timer Options


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 Refresh IntervalMinimum 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.




Media Settings

Table 4-1-6 Definition of Media Settings




Media Settings

Select codec from the drop down list. Codecs should be different for each Codec Priority.



Batch SIP Endpoint

If you want add batch Sip accounts, you can configure this page. Look out: this is only used when “This gateway registers with the endpoint” work mode.


Figure 4-2-1 Batch SIP Endpoint

Advanced SIP Settings


Table 4-3-1 Definition of Networking Options




UDP Bind Port

Choose a port on which to listen for UDP traffic.

Enable TCP

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

TCP Bind Port

Choose a port on which to listen for TCP traffic.

TCP Authentication Timeout

The maximum number of seconds a client has to authenticate. If the client

does not authenticate before this timeout expires, the client will be

disconnected.(default value is: 30 seconds).

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.



NAT Settings

Table 4-3-2 Definition of NAT Settings






Local Network


Format: or 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.





Advanced: NAT Settings

Table 4-3-3 Definition of NAT Settings Options




Start of RTP Port Range

Start of range of port numbers to be used for RTP.

End of RTP port Range

End of range of port numbers to be used for RTP.

RTP Timeout




Parsing and Compatibility



Table 4-3-4 Instruction of Parsing and Compatibility




Strict RFC Interpretation

Check header tags, character conversion in URIs, and multiline headers

for strict SIP compatibility(default is yes)

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.






Table 4-3-5 Instruction of Security




Match Auth Username

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


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.




Table 4-3-6 Instruction of Media




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




5. Network, Advanced and Logs


On “Network” page, there are “Network Settings”, “DDNS Settings”, and “Toolkit”.

Network Settings

There are three types of LAN port IP, Factory, Static and DHCP. Factory is the default type, and it is When you Choose LAN IPv4 type is “Factory”, this page is not editable.


A reserved IP address to access in case your gateway IP is not available. Remember to set a similar network segment with the following address of your local PC.


Figure 5-1-1 LAN Settings Interface


Table 5-1-1 Definition of Network Settings





The name of network interface.


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.


Physical address of your network interface.


The IP address of your gateway.


The subnet mask of your gateway.

Default Gateway

Default getaway IP address.

Reserved Access IP

A reserved IP address to access in case your gateway IP is not available.

Remember to set a similar network segment with the following address of

your local PC.


A switch to enable the reserved IP address or not.

ON(enabled), OFF(disabled)

Reserved Address

The reserved IP address for this gateway.

Reserved Netmask

The subnet mask of the reserved IP address.




Basically this info is from your local network service provider, and you can fill in four DNS servers.


Figure 5-1-2 DNS Interface


Table 5-1-2 Definition of DNS Settings




DNS Servers

A list of DNS IP address. Basically this info is from your local network service provider.


OpenVPN Settings

You can upload the OpenVPN client configuration, if success, you can see a VPN virtual network card on SYSTEM status page. About the configure format you can refer to the Notice and Sample configuration.


Figure 5-1-3 OpenVPN Interface

DDNS Settings

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


Figure 5-1-4 DDNS Interface


Table 5-1-3 Definition of DDNS Settings





Enable/Disable DDNS(dynamic domain name server)


Set the type of DDNS server.


Your DDNS account’s login name.


Your DDNS account’s password.

Your domain

The domain to which your web server will belong.





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


Figure 5-1-5 Network Connectivity Checking


Asterisk API

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



Figure 5-2-1 API Interface



Table 5-2-1 Definition of Asterisk API





Network port number

Manager Name

Name of the manager without space

Manager secret

Password for the manager.

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


If you want to deny many hosts or networks, use char & as separator.

<br/><br/>Example: or


If you want to permit many hosts or network, use char & as separator.

<br/><br/>Example: or


General information about the system and ability to run system management

commands, <br/>such as Shutdown, Restart, and Reload.


Information about channels and ability to set information in a running channel.


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


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


Permission to run CLI commands.  Write-only.


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


Permission to send and receive UserEvent.


Ability to read and write configuration files.


Receive DTMF events.  Read-only.


Ability to get information about the system.


Output of cdr, manager, if loaded.  Read-only.


Receive NewExten and Varset events.  Read-only.


Permission to originate new calls. Write-only.


Select all or deselect all.



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


Figure 5-2-2 Putty Access




Asterisk CLI

In this page, you are allowed to run Asterisk commands.



Figure 5-2-3 Asterisk Command Interface


Table 5-2-2 Definition of Asterisk API





Type your Asterisk CLI commands here to check or debug your




If you type “help” or “?” and execute it, the page will show you the executable commands.

Asterisk File Editor

On this page, you are allowed to edit and create configuration files.

Click the file to edit. 

Figure 5-2-4 Configuration Files List


Click “New Configuration File” to create a new configuration file. After editing or creating, please reload Asterisk.


On the “Log Settings” page, you should set the related logs on to scan the responding logs page. For example, set “System Logs” on like the following, then you can turn to “System” page for system logs, otherwise, system logs is unavailable. And the same with other log pages.


Figure 5-3-1 System Logs Control



Figure 5-3-2 System Logs Output


Notice: The same to Asterisk Logs and SIP Logs.


Table 5-3-1 Definition of LOG




System Logs

Whether enable or disable system log.

Auto clean

(System Logs)

switch on :

         when the size of log file reaches the max size,

         the system will cut a half of the file. New logs will be retained.<br>

switch off :

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

default on, max size=1MB.


Asterisk console verbose message switch.


Asterisk console notice message  switch.


Asterisk console warning message  switch.


Asterisk console debug message switch.


Asterisk console error message  switch.


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.












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