OpenVox DGW-100XR User Manual

DGW-100XR User Manual

Address: 10/F, Building 6-A, Baoneng Science and Technology Industrial Park, Longhua New

District, Shenzhen, Guangdong, China, 518048

Tel: [phone]

Fax: [phone]

Business Contact: [e-mail]

Technical Support: [e-mail]

Business Hours: 09:00-18:00(GMT+8) from Monday to Friday

URL: www.openvox.cn

DGW-100XR User Manual

OpenVox Communication Co.Ltd

Version1.1 (2015-8-20)

Full text

The overall layout adjustment

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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.

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Table of Contents

1.Overview................................................................................................................................. 5

What is DGW-100XR? 5

Sample Application 5

Product Appearance 5

Main Features 7

Physical Information 7

Software 8

2. System ................................................................................................................................... 9

Status 9

Time 10

Login Settings 11

General 12

Language Settings 12

Scheduled Reboot 12

Tools and Information 13

Reboot Tools 13

Update Firmware 13

Upload and Backup Configuration 14

Restore Configuration 14

Information 14

3. T1/E1 ................................................................................................................................... 16

General 16

ISDN-PRI 17

Advanced: Interface Type 17

ISDN: Signaling 18

SS7 20

Link Set Settings 20

Link Settings 21

SS7 Config. File Backup and Restore 22

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MFC/R2 22

Advanced: Interface Type 22

MFC/R2: Signaling 23

4.VOIP ...................................................................................................................................... 25

VOIP Endpoints 25

SIP Endpoints 25

Main Endpoint Settings 25

Advanced: Registration Options 28

Call Settings 28

Advanced Timer Settings 28

Advanced: Signaling Settings 29

IAX2 Endpoint 30

Advanced SIP Settings 33

Networking 33

Advanced: NAT Settings 34

Advanced: RTP Settings 35

Parsing and Compatibility 35

Security 36

Media 37

Codec Settings 37

Advanced IAX2 Settings 38

Advanced Fax Settings 40

5. Routing ................................................................................................................................ 42

Call Routing Rule 42

Groups 45

6. Network ............................................................................................................................... 47

WAN/LAN Settings 47

DDNS Settings 48

Toolkit 49

7. Advanced ............................................................................................................................. 50

Asterisk API 50

Asterisk CLI 51

Asterisk File Editor 52

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8. Logs ...................................................................................................................................... 54

System 57

Statistics 58

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1. Overview

What is DGW-100XR?

OpenVox T1/E1 Gateway is an open source asterisk-based VoIP Gateway solution for operators and
call centers. It is a converged media gateway product. This kind of gateway connects traditional
telephone system to IP networks and integrates VoIP PBX with the PSTN seamlessly. With friendly
GUI, users may easily setup their customized Gateway. Also secondary development can be
completed through AMI (Asterisk Management Interface). The DGW-100XR could support two
power supply and DGW-100X series gateway support one power supply.
It is developed with a wide selection of codecs and signaling protocol, including G.711A, G.711U,
G.729, G.722, G.723 and GSM. It supports PRI/SS7/R2 protocol. OpenVox T1/E1 Gateway has good
processing ability and stability and we provides 1/2/4 T1/E1 interface for your choice. The T1/E1
gateway will be 100% compatible with Asterisk, Elastix, trixbox, 3CX, FreeSWITCH SIP server and
VOS VoIP operating platform.

Sample Application

Figure 1-2-1Topological Graph

Product Appearance

The picture below is appearance of DGW-1004.

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Figure 1-3-1 Product Appearance

Figure 1-3-2 Front Panel

Table 1-3-1 Description of Front Panel

Interface

Function

Color

Work Status

1 Port 1-Port4

E1/T1 ports. The port numbers are different on different

models, from 1 to 4.

2 Reset

Reset button is used to restore the device.

3 RUN

Register indicator

Green

Slow blinking(Green 2s
and Flash 0.1s):Work
normally

Fast blinking(Green 0.5s
and Flash 0.5s): Work
abnormally

Fast blinking(Green 0.5s
and Flash 0.5s): Work
abnormally

No blinking: Dahdi Error

4 PWR

Power Status indicator

Green
On: Power is on

Off: Power is off

5 VGA

VGA monitor connector

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6 Eth1

Network interface

7 Eth0

Network interface

8 USB

USB interface

Figure 1-3-3 Backup Panel

The OpenVox DGW-100X series gateways provides one or two power supply, one power named DGW-100X ,the

other is named DGW-100XR, ‘R’ stands for Reduntant.

Main Features

 Based on Asterisk○R

 Editable Asterisk

○R

configuration file

 Wide selection of codecs and signaling protocol

 Support unlimited routing rules and flexible routing settings

 Stable performance, flexible dialing, friendly GUI

 Codecs support: G.711A, G.711U, G.729, G.723, G.722,GSM

 Support ports group management

 Echo Cancellation

 Connect legacy PBX systems to low-cost VoIP services

 Connect legacy PBX systems to remote sites over private VoIP links

 Connect IP PBX systems to legacy TDM services

Physical Information

Table 1-5-1 Description of Physical Information

Weight

4301g(VS-GW1600-20G)

Size

44cm*30cm*4.5cm

Temperature

-20~70°C (Storage)

0~40°C (Operation)

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Operation humidity

10%~90% non-condensing

Max power

46W

LAN port

1

WAN port

1

Software

Default IP: 172.16.100.1
Username: admin
Password: admin

Notice: Log in

Figure 1-6-1 LOG IN Interface

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2. System

Status

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

Figure 2-1-1 System Status

Table 2-1-1 Description of System Status

Options

Definition

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Interface Status

Show the status of port, include "OK" and ”Down”. "Down"
means no trunk line connected; "OK" means the trunk line of
port is available.

Channels Status

Show the Channels status of port, include "Idle". "Busy".

"Disable" and “S channel”.
"Idle" means it is available;
"Busy" means the channel is busy;
"Disable" means it is unavailable;

“S channel” means signaling channel.

Time

Table 2-2-1Description 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 timezone strings.

NTP Server 1

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

NTP Server 2

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

NTP Server 3

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

Auto-Sync from NTP

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

Sync from NTP

Sync time from NTP server.

Sync from Client

Sync time from local machine.

For example, you can configure like this:

Figure 2-2-1 Time Settings

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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 “Web Login Settings” and “SSH Login Settings”. If you have changed these settings, you
don’t need to log out, just rewriting your new user name and password will be OK. Also you can
specify the web server port number.

Table 2-3-1Description of Login Settings

Options

Definition

User Name

Your gateway dose not have administration role.
All you can do here is defining the user name 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

Password

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

Confirm
Password

Please input the same password as 'Password' above.

Port

Specify the web server port number.

Figure 2-3-1 Login Settings

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

General

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”.

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

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

Tools and Information

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

Reboot Tools

You can choose system reboot and Asterisk reboot separately.

Figure 2-5-1 Reboot Prompt

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

Table 2-5-1 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.

Update Firmware

We offer 2 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, if you choose that, you will see

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some information below.

Figure 2-5-2Prompt Information

Upload and Backup Configuration

If you want to update your system and remain your previous configuration, you can first backup

configuration, then you can upload configuration directly. That will be very convenient for you.

Figure 2-5-3 Upload and Backup

Restore Configuration

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

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

reset to the factory status.

Figure 2-5-4 Factory Reset

Information

On the “Information” page, there shows some basic information about the T1/E1 gateway. You can

see software and hardware version, storage usage, memory usage and some help information.

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Figure 2-5-5 System Information

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3. T1/E1

General

Figure 3-1-1 General Settings

Table 3-1-1 Definition of General Settings

Figure 3-1-2 Port Details

Table 3-1-3 Definition of Port Details

Local

Your local. This will be used for the tone style. Used when in-call
indications need to be generated such as ring back, busy,
congestion, and other call-oriented inband tone signals.

Interface Type

It shows you the current type of port. It has two type:E1 and T1

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Options

Definition

Timing Source

Timing Source indicate the ports as to which should be used to recover
the clock.(0 for master mode, upper for client mode, small number
have higher priority )

Interface

Choose a line type for this interface, all ports must be the same type.

Framing

Framing method for this interface

Coding

Coding method for this interface

Line Build-out

Line build-out represents the length of the cable form the port on this
gateway to the next device.

CRC4

Enable cyclic redundancy checking for error checking on line. CRC-4
support is required for all network switches in Europe, but many older
switches and PBXs don’t support it.

Signaling

It shows you what signaling the port uses.

Switch Type

Only used for PRI

Description

An optional description of this interface to be used for reference only.

ISDN-PRI

Advanced: Interface Type

Figure 3-2-1 Advanced: Interface Type

Table 3-2-1Definition of Interface Type

Options

Definition

Echo Cancellation

Whether or not to enable echo cancellation on this line

RX Gain Whole
number

-24 to 24 and

Gain for the rx (receive -into Asterisk)channel.Default:0.0

TX Gain Whole
number

-24 to 24 and

Gain for the tx(transmit -out of Asterisk
Asterisk)channel.Default:0.0

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ISDN: Signaling

Figure 3-2-2 ISDN: Signaling

Table 3-2-2 Definition of Signaling

Options

Definition

Q.SIG Channel Mapping

Sets logical or physical channel mapping. In logical channel mapping,
channels are mapped to 1-30.In physical channel mapping, channels
are mapped to 1-15,17-31,skipping the number used for the data
channel, Default is physical.

Enable Caller ID

Whether or not to use caller ID

PRI Dial Plan for Dialed
Number

PRI Dialplan: The ISDN-levei Type of Number (TON) or numbering
plan, used for the dialed number. Leaving this as ‘unknown’ (the
default) works for most cases. In some very unusual circumstances,
you may need to set this to; ’dynamic’ or ‘redundant’

PRI Dial Plan for Dialing
Number

PRI Local Dialplan: Only RARELY used for PRI(sets the calling number’s
numbering plan).In North America, the typical use is sending the 10

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digit; caller ID number and setting the prilocaldialplan to ‘national’
(the default).Only VERY rarely will you need to change this.

Network Specific Facility
(NSF) Messages

Some switches (AT&T especially) require network specific facility IE.
Supported values are currently ‘none’,’sdn’,’ megacom’,’
tollfreemgacom’, ’ account’

Idle Bearer Reset

Whether or not to reset unused B channels

Idle Bearer Reset Period

Sets the time in seconds between restart of unused B channels;
defaults to ‘never’

Display Send

Send/receive ISDN display IE options, the display options are a
comma separated list of the following options:
Block:
Do not pass display text data.
Name_ initial:
Use display text in SETUP/CONNECT messages as the party name.
Name_ update:
Use display text in other messages (NOTIFY/FACILITY)for COLP name
update.
Name:
Combined name_ initial and name_ update options.
Text:
Pas any unused display text data as an arbitrary display message
during a call. Sent text goes out in default to ‘name’

Display Receive

Send/receive ISDN display IE options. The display options are a
comma separated list of the following options:
Block:
Do not pass display text data.
Name_ initial:
Use display text in SETUP/CONNECT messages as the party name.
Name_ update:
Use display text in other messages (NOTIFY/FACILITY) for COLP name
update.
Name:
Combined name_ initial and name_ update options.
Text:
Pas any unused display text data as an arbitrary display message
during a call. Sent text goes out in default to ‘name’

Overlap Dialing

Enable overlap dialing mode--sending overlap digits.

Allow Progress When Call
Released

Allow inband audio (progress) when a call is DISCONNECT Ted by the
end of a PRI

Out-of-Band Indications

PRI Out of band indications. Enable this to report Busy and congestion
on a PRI using out_ of_ band notification. Inband indication, as used
by the gateway doesn’t seem to work with all telcos.

Facility-based ISDN

To enable transmission of facility-based ISDN supplementary services

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Supplementary Services

(such as caller name form CPE over facility), enable this option.
Cannot be changed on a reload.

Exclusive Channel
Selection

If you need to override the existing channels selection routine and
force all PRI channels to be marked as exclusively selected, set this to
yes. priexclusive cannot be changed on a reload.

Ignore Remote Hold
Indications

If you wish to ignore remote hold indications (and use MOH that is
supplied over the B channel) enable this option.

Block Outbound Caller ID
Name

Enable if you need to hide just the name and the number for legacy
PBX use. Only applies to PRI channels.

Wait for Caller ID Name

Support caller ID on call waiting

SS7

Link Set Settings

Figure 3-3-1 Link Set Settings

You can click button as shown below, when there are several link set, only one can be set to

the default.

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Figure 3-3-2 SS7 Link Set Settings

Table 3-3-1 Definition of SS7 Link Set Settings

options

Definition

Name

The linkset’s name

Enabled

The linkset is enable or disable

Enabled_ st

The end_of_pulsing (ST) is not used to determine when incoming address is
complete

Use Connect

Reply incoming call with CON rather than ACM and ANM

Hunting Policy

The CIC hunting policy (even_mu, odd_lru, seq_lth, seq_htl) is even CIC
numbers, most recently used

Subservice

The subservice field: national (8), international l(0), auto or decimal/hex
value; The auto means that the subservice is obtained from first received
SLTM.

t35

The value and action for t35. Value is in msec, action is either st or timeout;
if you use overlapped dialing dial plan, you might choose:t35=>4000,st

variant

Running under SS7 standard

OPC

The point code for this SS7 signaling point

DPC

The destination point (peer) code

Set to Default

Set the linkset as the default linke set

Link Settings

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Figure 3-3-3 Link Settings

You can click button as shown below.

Figure 3-3-4 SS7 Link Settings

SS7 Config. File Backup and Restore

Figure 3-3-5 Config. File Backup and Restore

MFC/R2

Advanced: Interface Type

Figure 3-4-1 Advanced: Interface Type

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Table 3-4-1 Definition of Interface Type

options

Definition

Echo Cancellation

Whether or not enable echo cancellation on this line

RX Gain Whole number -24
to 24 and multiple of 3

Gain for the rx (receive_ into Asterisk) channel.
Default:0.0

TX Gain Whole number -24
to 24 and multiple of 3

Gain for the tx (receive_ into Asterisk) channel.
Default:0.0

MFC/R2: Signaling

Figure 3-4-2 MFC/R2: Signaling

Table 3-4-2Definition of MFC/R2: Signaling

options

Definition

Enable Caller ID

Whether or not to use caller ID

mfcr2_init_cas_bit

The initial position of the CAS bits (also known as ABCD bits)

mfcr2_get_ani_first

Whether or not to get ANI before getting DINS; some telcos require ANI first
some others do not care; if this go wrong, change this value

mfcr2_max_ani

Max amount of ANI to ask for

mfcr2_max_dnis

Max amount of DNIS to ask for

mfcr2_category

Usually national-subscriber works just fine; you can change this setting from
the dialplan ; by setting the variable MFCR2-CATEGORY;(remembering ti set­MFCR2-CATEGORY from originating channels);MFCR2-CATEGORY will also be
a variable in your context; on incoming calls set to the value received from

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the far end;mfcr2-category=national-subscriber

mfcr2_mfback_time
out

MFC/R2 value in milliseconds for the MF timeout

mfcr2_metering_pul
se_timeout

MFC/R2 value in milliseconds for the metering pulse timeout

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4.VOIP

VOIP Endpoints

SIP Endpoints

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

Figure 4-1-1 SIP Status

Main Endpoint Settings

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

existed endpoints, you can click button.

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.)

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Figure 4-1-2 None 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 Endpoint Register with Gateway

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

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Figure 4-1-4 This Gateway Register with the Endpoint

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 end point use to authenticate with the

gateway

Password

Password the endpoint will use to authenticate with the

gateway. Allowed characters

Registration

Whether this endpoint will registers with this gateway.

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.

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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

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

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

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).

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.

Caller ID Presentation

Whether or not to display Caller ID.

Advanced Timer Settings

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Table 4-1-4 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

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

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 1800s.
Session Refresher

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

Advanced: Signaling Settings

Table 4-1-5Definition of Signaling Options

Options

Definition

Progress Inband

If we should generate in-band ringing. Always use ‘never’ to never use in­band signalling,
Even in cases where some buggy devices might not render it. Valid values:
yes, no, never. Default: never.

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 changes. Turn This
option off to force the SDP session version number and treat all SDP data as
new data. This is require for devices that send non-standard SDP packets
(observed with Microsoft OC S).By default
This option is on.

Allow Transfers

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

Allow

Whether or not to allow 302 or REDIR to non-local SIP address .Note that

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Promiscuous
Redirects

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 100 Trying when the endpoint registers.

Table 4-1-6 Definition of Fax Options

Options

Definition

Mode

Working mode T.38 and T.30

Enabled

Enabled

Error Correction

Error Correction

Max Datagram

In some cases,T.38 endpoints will provide a T38FaxMxDatagram value (during
T.38 setup) that is based on an incorrect interpretation of the T.38
recommendation, and result in failures because Asterisk does not believe it can
send T.38 packets of a reasonable size to that endpoint (Cisco media gateway
are one example of this situation).In these cases, during a T.38 call you will see
warring messages on The console/in the logs from the Asterisk UDPTL stack
complaining about lack of buffer space to send T.38FaxMaxDatagram value
specified by the other end[point, and use a configured value instead.

Fax Detect

FAX detection will cause the SIP channel to jump to the ‘faX’ extension (if exists)
based one or more events being detected. The events that can be detected are
an incoming CNG tone or an incoming T.38 re-INVITE request.

Fax Activity

activate T38 fax gateway with ‘timeout’ seconds

Fax Timeout

activate T38 fax gateway with ‘timeout’ seconds

IAX2 Endpoint

Figure 4-1-5 IAX2 Endpoint

You can click button as shown below

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Figure 4-1-6 Edit IAX Endpoint "9001"

Table 4-1-6 Definition of IAX2 Endpoint

Options

Definition

Name

A name which is able to read by human.

And it’s only used for user’s reference.

User name

User name the endpoint will use to authenticate with the gateway

Password

Password the endpoint will use to authenticate with gateway.

Allowed characters

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Registration

Whether this endpoint will register to this gateway or 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.

Auth

Authentication method for connections

Transfer

Disable or not IAX2 native transfer

Trunk

Use IAX2 trunking with this host

Qualify

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

Qualify Smothing

Use an average of the last two PONG result to reduce falsely detected LAGGED

host. The default is ‘no’.

Qualify Freq Ok

How frequently to ping the peer when everything seems to be OK, in milliseconds.

Qualify Freq not

Ok

How frequently to ping the peer when it’s either;

LAGGED or UNAVAILABLE, in milliseconds.

Port

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

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.

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

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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).

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

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.

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Advanced: NAT Settings

Table 4-2-2 Definition of NAT Settings Options

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

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 exeternaddr 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 of
allowing your users to register at the same address as a SIP provide.

Externally Mapped
TCP Port

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

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 SDP messages. Examples: External address=12.34.56.78
External address=12.34.56.78.9900

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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.

Advanced: RTP Settings

Table 4-2-3 Definition 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.

Parsing and Compatibility

Table 4-2-4 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.

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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 endpoint
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 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

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

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 4-2-5 Instruction of Security

Option

Definition

Match Auth
Username

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

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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 4-2-6 Instruction of Media

Options

Definition

TOS for SIP
Packets

Sets type of service for SIP packets

TOS for RTP
Packets

Sets type of service for RTP packets

Codec Settings

Select codecs from the list below.

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Figure 4-2-1 Codec Settings

Advanced IAX2 Settings

Table 4-3-1 Instruction of General

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

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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.

Table 4-3-2 Instruction of Music on Hold

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 fallback to English(en)

Table 4-3-3 Instruction of Codec Settings

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.

Table 4-3-4 Instruction of Jitter Buffer

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.

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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.

Table 4-3-5 Instruction of Misc Settings

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.

Table 4-3-6 Instruction of Quality of Service

Options

Definition

Tos

Type of service

Cos

Class of service

Advanced Fax Settings

Table 4-4-1 Instruction of Quality of Fax Settings

Options

Definition

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Udptl Start

DPTL start configure addresses

Udptl End

DPTL end configure addresses

Udptl Checksums

Whether to enable or disable UDP checksums on UDPTL traffic

Udptl Fec Entries

The number of error correction entries in a UDPTL packet

Udptl Fec Span

The span over which parity is calculated for FEC in a UDPTL packet

Use Even Ports

Some VoIP providers will only accept an offer with an even-numbered
UDPTL port. Set this option so that Asterisk will only attempt to use
even-numbered ports when negotiating T.38. Default is no.

Maximum
Transmission Rate

Maximum Transmission Rate

Minimum Transmission
Rate

Minimum Transmission Rate

Send Progress/Status
events to manager
session

Manager events with ‘call’ class permissions will receive events
indicating the steps to initiate a fax session. Fax completion events are

always sent to manager sessions with ‘call’ class permissions,

regardless of the value of this option.

Modem Capabilities

Set this value to modify the default modem options.
Defasult:v17,v27,v29

ECM

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

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5. Routing

Figure 5-1-1 Routing Rules

You are allowed to set up new routing rule by , and after setting routing

rules, move rules’ order by pulling up and down, click button to edit the routing and to

delete it. Finally click the button to save what you set. shows current routing

rules. Otherwise you can set up unlimited routing rules.

Call Routing Rule

There is an example for Routing rules number conversion, it transform calling, called number at the
same time. Suppose you want eleven numbers start at 159 to call the eleven numbers of start at

136. Calling transform delete the three numbers from left, then writing number 086 as prefix,
delete the last four numbers, and then add number 0755 at the end, it will show caller name is
openvox. Called transform adds 086 as prefix , and Change the last two number to 88.

Figure 5-1-2

processing

rules

prepend

prefix

Match

pattern

SdfR

StA

RdfR

Caller

Name

Calling Transf
ormation

086

159

×××

×××

××

4

0755

OpenVox

Called

transformation

086

136

×××

×××

××

2

88 N/A

You can click button to set up your routings.

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Figure 5-1-3 Example of Set Up Routing Rule

The figure above realizes that calls from “support” SIP endpoint switch you have registered will be
transferred to Port-1. When “Call Comes in From” is 1001, “prepend”, “prefix” and “match pattern”
in “Advanced Routing Rule” are ineffective, and just “CallerID” option is available.

Table 5-1-1 Definition of Routing Options

Options

Definition

Routing Name

The name of this route. Should be used to describe what types of
calls this route matches (for example, 'SIP2Ports' or 'Ports2SIP').

Call Comes in From

The launching point of incoming calls.

Send call Through

The destination to receive the incoming calls.

Table 5-1-2 Description of Advanced Routing Rule

Options

Definition

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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 match.

If 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 match.
The 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
SDfR(Stripped Digits from Right): The amount of digits to be
deleted from the right end of the number. If the value of this
item exceeds the length of the current number, the whole
number will be deleted.
RDfR( Reserved Digits from Right) :Designated information to be
added to the right end of the current number.
StA(Suffix to Add):Designated information to be added to the
right end of the current number.
Caller Name: What caller name would you like to set before
sending this call to the endpoint. Native language charset is
allowable, e.g. Chinese charset, Latin charset.

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.

You can create various time routes and use these time conditions to limit some specific calls.

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Figure 5-1-4Time Patterns that will use this Route

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

Figure 5-1-5Change Rules

You can set your caller ID name and caller number as you like before sending the call to the
endpoint. You can also configure forward number when you have a transfer call.

Figure 5-1-6 Failover Call Through Number

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

Groups

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

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Figure 5-2-1 Establish Group

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6. Network

On “Network” page, there are three sub-pages, “WAN Settings”, “DDNS Settings”, and “Toolkit”.

WAN/LAN Settings

There are two types of WAN port IP, Static and DHCP. Static is the default type, and it is

172.16.100.1. The LAN port is a fixed IP and it is 192.168.100.1.

Figure 6-1-1 WAN/LAN Settings Interface

Table 6-1-1Definition of WAN/LAN Settings

Options

Definition

Interface

The name of network interface.

Type

The method to get IP.
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.

Network

The subnet mask of your gateway.

Default Gateway

Default getaway 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

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

DDNS Settings

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

Figure 6-2-1 DDNS Interface

Table 6-2-1 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.

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Toolkit

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

Figure 6-3-1 Network Connectivity Checking

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7. Advanced

Asterisk API

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

Figure 7-1-1 API Interface

Table 7-1-1 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.

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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.

Dialplan

Receive NewExten and Var Set 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 172.16.100.110/255.255.0.0 is allowed to access the
gateway API. Please refer to the following figure to access the gateway API by putty.

172.16.100.110 is the gateway’s IP, and 5038 is its API port.

Figure 7-1-2 Putty Access

Asterisk CLI

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

Figure 7-2-1 Asterisk Command Interface

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Table 7-2-1 Definition of Asterisk CLI

Options

Definition

Command

Type your Asterisk CLI commands here to check or debug
your gateway.

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.

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Figure 7-3-1 Configuration Files List

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

reload Asterisk.

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8. Logs

On the “Log Settings” page, you should set the related logs on to scan the responding logs page.
For example, set “System Logs” on like the following, then you can turn to “System” page for

system logs, otherwise, system logs is unavailable. And the same with other log pages.

Log Settings

Figure 8-1-1 Logs Settings

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Figure 8-1-2 System Logs Output

Table 8-1-1 Definition of Logs

Options

Definition

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, default 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, default size=100KB

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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=2MB

IAX2 Logs

Whether enable or disable IAX log

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=2MB

MFC/ R2 Logs

Whether enable or disable MFC/ R2 Logs log.

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=100KB

PRI Logs

PRI port logs. You can choose one or more ports. If you
choose "All", the "PRI" page will show you the logs about all
the ports.

Auto clean (PRI 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=2MB

.SS7 Logs

Whether enable or disable SS7 log

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=100KB

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Call Statistics

Whether enable or disable Call Statistics.

System

Figure 8-2-1 System Logs

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OpenVox Communication Co.Ltd

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Asterisk

Figure 8-3-1 Asterisk Logs

On the pages of “system”, ”Asterisk”, “SIP”, “IAX2”, “SS7”, and “MFC/R2”, there are some
functions: Displays the log by port, refresh regularly and log download.

Statistics

Figure 8-9-1 Call Statistics

The figure of call statistics, you’ll find “Answered” “congestion” “busy” “failed” “no answer”
“current calls” “accumulated” “calls duration” .