iAG804/iAG808 User Manual

                               

 

Copyright

Copyright© 2013 OpenVox Inc. All rights reserved. No part of this document may be reproduced without prior written permission.

 

Confidentiality

Information contained herein is of a highly sensitive nature and is confidential and proprietary to OpenVox Inc. No part may be distributed, reproduced or disclosed orally or in written form to any party other than the direct recipients without the express written consent of OpenVox Inc.

 

Disclaimer

OpenVox Inc. reserves the right to modify the design, characteristics, and products at any time without notification or obligation and shall not be held liable for any error or damage of any kind resulting from the use of this document.

OpenVox has made every effort to ensure that the information contained in this document is accurate and complete; however, the contents of this document are subject to revision without notice. Please contact OpenVox to ensure you have the latest version of this document.

 

Trademarks

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

 

1. Overview

What is iAG804/808?

This document is to explain the quad-FXO module of analog gateway.

OpenVox 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 four models with Analog Gateway, the 4FXS, 8FXS, 4FXO and 8FXO, and there are 4/8 ports in iAG804/808. The Modular Design Analog Gateways are 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 to quickly reduce communication expenses and maximize cost-savings.

The Analog gateway use standard SIP protocol and compatible with Leading IMS/NGN platform, IPPBX and SIP servers, support most of the VoIP operating platforms such as Asterisk, Elastix, 3CX, FreeSWITCH ,Broadsoft etc.

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

 

Figure 1-3-2 Front Panel

1 : System LED

2 : Network interface LED

3 : Power Indicator

 

Figure 1-3-4 Back Panel

1 : Analog Telephone Interface (8)

2 : Channel indicator (8)

3 : USB Interface(1)

    4 : Ethernet ports (1)

    5 : Fan vent

    6 : Power socket

    7 : Reload button

Main Features

  • Modular 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

 

Weight

580g

Size

21cm*21cm*3.6cm

Temperature

-20~70°C (Storage)

0~40°C (Operation)

Operation humidity

10%~90% non-condensing

Power source

12V DC/4A

Max power

16W

LAN port

1

 

 

Software

Default IP: 172.16.99.1

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

Status

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

Figure 2-1-1 System Status

Time

Table 2-2-1 Description of Time Settings

 

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].

Auto-Sync from NTP

Whether enable automatically synchronize from NTP server or not. ON is enable, OFF is disable this function.

 

 

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

 

Options

Definition

User Name

NOTES: Your gateway doesn’t have administration role.

All you can do here is defining the username and password to manage

your gateway. And it has all privileges to operate your gateway.

User Name: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.

 

 

Figure 2-3-1 Login Settings

 

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

General, Cluster, 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.

Figure 2-4-3 Snmp Agent

Figure 2-4-4 Working Mode

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 2-4-5 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

 

Options

Definition

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 2-4-6 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 2-4-7 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 2-4-8 Factory Reset

Information

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 2-4-9 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.

Advanced Settings

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

Global Settings

Figure 3-2-1 General Configuration

Table 3-2-1 Instruction of General

 

Options

Definition

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.

Codec

Set the global encoding : mulaw, alaw.

Impedance

Configuration for impedance.

Echo cancel tap length

Hardware echo canceler tap length.

VAD/CNG

Turn on/off VAD/CNG.

Flash/Wink

Turn on/off Flash/wink.

Min flash time

Min flash time.(in milliseconds).

Max flash time

Max flash time.(in milliseconds).

Hangup on polarity switch

Turn on/off hide caller id function.

 

 

Figure 3-2-2Hardware Gain

 

Table 3-2-2 Instruction of Hardware gain

 

Options

Definition

FXO Rx gain

Set the FXO port Rx gain. Range: from -150 to 120.

 

 

 

 

 

 

 

FXO Tx gain

Set the FXO port Tx gain. Range: from -150 to 120. 

 

 

 

Figure 3-2-3 CallerID detect

 

Table 3-2-3 Instruction of CallerID detect

 

Options

Definition

Use Callerid

Turn on/off callerid detect function

Hide Callerid

Turn on/off callerid detect function

 

Figure 3-2-4 Busy detect

 

Table 3-2-4 Instruction of Busy detect

 

Options

Definition

Busy detect

Turn on/off busy detect function

Busy count

How many busy tones to wait for before hanging up. The default is 3, but it might be safer to set to 6 or even 8.

Busy country

Set the busy detect country.

 

 

 

Figure 3-2-5 Silence detect

 

Table 3-2-5 Definition of Silence detect

 

Options

Definition

Silence detect

Turn on/off silence detect function

 

 

 

 

 

 

 

Silence threshold

What we consider silence: the lower, the more sensitive, eg:250 is 250ms. Range: 100 to 500(100 to 500ms), default: 250

Max silence

How many silence threshold of silence before hanging up(eg: 16 is 250ms*16=4s). Range: 2 to 1020 (200ms to 512s), default: 80(20s)

Rx threshold

Range:-20 dBm0 to -40 dBm0, default: 20(-20 dBm0), all values are understood to be negative.

Tx threshold

Range:-20 dBm0 to -40 dBm0, default: 20(-20 dBm0), all values are understood to be negative.

 

 

Figure 3-2-6 Fax Configuration

 

Table 3-2-6 Definition of Fax

 

Options

Definition

Mode

Set the transmission mode.

 

 

 

 

 

 

 

Rate

Set the rate of sending and receiving.

Ecm

Enable/disable T.30 ECM (error correction mode) by default.

 

 

Figure 3-2-7 Country Configuration

 

Table 3-2-7 Definition of Country

 

Options

Definition

Country

Configuration for location specific tone indications.

Ring cadence

List of durations the physical bell rings.

Dial tone

Set of tones to be played when one picks up the hook.

Ring tone

Set of tones to be played when the receiving end is ringing.

Busy tone

Set of tones played when the receiving end is busy.

Call waiting tone

Set of tones played when there is a call waiting in the background.

Congestion tone

Set of tones played when there is some congestion.

Dial recall tone

Many phone systems play a recall dial tone after hook flash.

Record tone

Set of tones played when call recording is in progress.

Info tone

Set of tones played with special information messages (e.g., number is out of service.)

Stutter tone

 

 

Special Function Keys

Figure 3-3-1 Function keys

 

Table 3-3-1 Definition of Function keys

 

Options

Definition

None Keys Blind Transfer

None Keys Blind Transfer help.

 

 

 

 

 

 

 

Blind Transfer

Blind Transfer help.

Asked Transfer

Asked Transfer help.

 

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

Name

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

Username

User Name the endpoint will use to authenticate with the gateway.

Password

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

characters.

Registration

Whether this endpoint will register to this gateway ro this gateway to the endpoint.

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.

Notice: if the input here is hostname and your DNS has changed, you must reboot asterisk.

Backup 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.

Notice: if the input here is hostname and your DNS has changed, you must reboot asterisk.

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

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

SUBSCRIBE for MWI

Whether or not to subscribe to receive the MWI.

VOS Encryption

Whether or not to enable VOS Encryption.

 

 

Advanced: Registration Options

Table 4-1-2 Definition of Registration Options

 

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.

Quality

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 signaling instead of sending signaling directly to endpoints.

 

 

Call Settings

Table 4-1-3 Definition of Call Options

 

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

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 P-Asserted-Identity.

Caller ID Presentation

Whether or not to display Caller ID.

 

 

Advanced: Signaling Settings

Table 4-1-4 Definition of Signaling Options

 

Options

Definition

Progress Inband

If we should generate in-band ringing.

Always use ‘never’ to never use in-band signaling, even in cases where some buggy devices might not render it.

 Valid values: yes, no never.

Default: 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 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 Forwards

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

Send TRYING on REGISTER

Send a 100 Trying when the endpoint registers.

 

 

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 Interval

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.

 

 

Media Settings

Table 4-1-6 Definition of Media Settings

 

Options

Definition

Media Settings

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

 

 

 

Advanced SIP Settings

Networking

Table 4-2-1 Definition of Networking Options

 

Options

Definition

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 4-2-2Definition of NAT Settings

 

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 staticly defined hosts. This helps avoid the configuration error for 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 Address

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

External Address=hostname :port] specifies a static address[:port] to be used in SIP and messages. Examples:

External Address=12.34.56.78

External Address=12.34.56.78:9900

External Hostname

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

External Hostname= hostname [:port] is similar to External Address. Examples: External Hostname = foo.dyndns.net.

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.

 

 

RTP Settings

Table 4-2-3Definition of RTP Settings Options

 

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 range of port numbers to be used for RTP.

RTP Timeout

RTP Timeout

 

 

Parsing and Compatibility

Table 4-3-4 Instruction of Parsing and Compatibility

 

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

When a dialog is started with another SIP endpoint, the other endpoint should include an Allow header telling us what SIP methods the endpoint implements. However, some endpoints either do not include an Allow header or lie about what methods they implement. In the former case, the gateway makes the assumption that the endpoint support all known SIP methods. If you know that your SIP endpoint does not provide support for a specific method, then you may provide a list of methods that your endpoint does not implement in the disallowed_methods option. Note that if your endpoint is truthful with its Allow header, then there is no need to set this option.

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.

 

 

Security

Table 4-3-5 Instruction of Security

 

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.

 

 

 

 

 

Media

Table 4-3-6 Instruction of Media

 

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

 

5.Routing

Call Routing Rules

Figure 5-1-1 Routing Rules

 

You are allowed to set up new routing 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  shows current routing rules. Otherwise you can set up unlimited routing rules.

Figure 5-1-2 Create a Call Routing Rule

Figure 5-1-2 Modify a Call Routing Rule

 

Table 5-1-1 Definition of Call Routing Rule

 

 

Options

Definition

Routing Name

The name of this route. Should be used to describe what types of calls this route matches (for example, ‘SIP2GSM’ or ‘GSM2SIP’).

Call Comes in From

The launching point of incoming calls.

Send Call Through

The destination to receive the incoming calls.

 

 

Figure 5-1-3 Advance Routing Rule

 

Table 5-1-2 Definition of Advance Routing Rule

 

Options

Definition

Dial Patterns that will use this Route

Callee ID Manipulation Help

Set the Caller ID Name to

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

Set the Caller ID Number to

What caller number would you like to set before sending call to the endpoint.

Forward Number

What destination number will you dial?

This is very useful when you have a transfer call.

Failover Call Through Number

The gateway will attempt to send the call out each of these in the order you specify.

 

 

Groups

Figure 5-2-1 Group Rules

 

You can click button to set new group, and if you want to modify existed group, you can click  button.

Figure 5-2-2 Create a Group

 

Figure 5-2-2 Modify a Group

 

Table 5-2-1 Definition of Routing Groups

 

Options

Definition

Group Name

The mean of this route. Should be used to describe what types of calls this route match (for example, ‘sip1 TO port1’ or ‘port1 To sip2’).

 

 

6 Network, Advanced and Log

Network

On “Network” page, there are “Network Settings”, ”VPN Setting”, “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 172.16.99.1. 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.

Figure 6-1-1 LAN Settings Interface

 

 

Table 6-1-1 Definition of Network Settings

 

Options

Definition

Interface

The name of network interface.

Type

The method to get IP.

Factory: Getting IP address by Slot Number (System à 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.

Netmask

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.

Enable

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 6-1-2 DNS Interface

 

Table 6-1-2 Definition of DNS Settings

 

Options

Definition

DNS Servers

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

 

VPN Settings

You can upload the VPN 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 6-1-3 OpenVPN Interface

Figure 6-1-4 PPTP VPN Interface

DDNS Settings

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

 

 

 

Figure 6-1-5 DDNS Interface

 

Table 6-1-3 Definition of DDNS Settings

 

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.

 

 

Toolkit

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

Figure 6-1-6 Network Connectivity Checking

 

Advanced

Asterisk API

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

Figure6-2-1 API Interface

 

Table 6-2-1 Definition of Asterisk API

 

Options

Definition

Port

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.

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

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

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

Call

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

Log

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

Verbose

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

Command

Permission to run CLI commands.  Write-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.

Dialplan

Receive NewExten and VarSet events.  Read-only.

Originate

Permission to originate new calls. Write-only.

All

Select all or deselect all.

 

 

Once you set like the above figure, the host 172.16.123.123/255.255.0.0 is allowed to access the gateway API. Please refer to the following figure to access the gateway API by putty. 172.16.123.123 is the gateway’s IP, and 5038 is its API port.

 

Figure 6-2-2 Putty Access

Asterisk CLI

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

Figure 6-2-3 Asterisk Command Interface

 

Table 6-2-2 Definition of Asterisk API

 

Options

Definition

Command

Type your Asterisk CLI commands here to check or debug your gateway. e.g, type “help” or “?” you will get all help information.

 

 

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 6-2-4 Configuration Files List

 

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

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. 

 

Figure 6-3-1 System Logs Control

Figure 6-3-2 System Logs Output

Notice: The same to Asterisk Logs and SIP Logs.

 

 

 

Table 6-3-1 Definition of Log Setting

 

Options

Definition

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.

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.

Call Detail Record

Displaying Call Detail Records for channel.

Auto clean

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=20MB.

 

 

CDR

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

                                                                                                                 

Figure 6-3-3 CDR Output

 

 

 

 

 

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