Analog Gateway User Manual
OpenVox Communication Co. ,LTD.
1
OpenVox Communication Co Ltd
Analog Gateway User Manual
Version 2.0
Analog Gateway User Manual
OpenVox Communication Co. ,LTD.
2
OpenVox Communication Co Ltd
Address: 10/F, Building 6-A, Baoneng Science and Technology Industrial Park, Longhua New District, Shenzhen,
Guangdong,China 518109
Tel: +86-755-66630978, 82535461, 82535362
Business Contact: sales@openvox.cn
Technical Support: support@openvox.cn
Business Hours: 09:00-18:00(GMT+8) from Monday to Friday
URL: www.openvox.cn
Thank You for Choosing OpenVox Products!
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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.
Statement
This document applies to all VoxStack series analog gateways, including FXS/FXO/Hybrid. Different
types of gateways may have functional differences. For details, please contact OpenVox sales or
technical support.
Trademarks
All other trademarks mentioned in this document are the property of their respective owners.
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Revise History
Version
Release Date
Description
1.0
22/12/2017
First Version
2.0
23/7/2019
Upgraded 2.0 version
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Contents
1. Overview .......................................................................................................................................... 8
1.1 What is Analog Gateway Series? ............................................................................................ 8
1.2 Sample Application ................................................................................................................ 9
1.3 Product Appearance ............................................................................................................. 10
1.4 Main Features ...................................................................................................................... 11
1.5 Physical Information ............................................................................................................. 11
1.6 Software ............................................................................................................................... 13
2. System ............................................................................................................................................ 14
2.1 Status .................................................................................................................................... 14
2.2 Time ...................................................................................................................................... 14
2.3 Login Settings ....................................................................................................................... 15
2.4 General ................................................................................................................................. 17
2.4.1 Language Settings ..................................................................................................... 17
2.4.2 Scheduled Reboot ..................................................................................................... 17
2.4.3 SNMP Agent .............................................................................................................. 18
2.5 Tools ..................................................................................................................................... 18
2.6 Information .......................................................................................................................... 19
2.7 Config Record ....................................................................................................................... 20
2.8 Config Label .......................................................................................................................... 20
3. Analog ............................................................................................................................................ 21
3.1 Channel Settings ................................................................................................................... 21
3.2 Pickup Settings ..................................................................................................................... 22
3.3 Dial Matching Table .............................................................................................................. 23
3.4 Advanced Settings ................................................................................................................ 24
3.5 Special Function Keys ........................................................................................................... 28
3.6 FXO ....................................................................................................................................... 29
3.7 FXS ........................................................................................................................................ 30
3.8 Driver .................................................................................................................................... 31
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4. SIP ................................................................................................................................................... 33
4.1 SIP Endpoints ........................................................................................................................ 33
4.1.1 Main Endpoint Settings ............................................................................................. 33
4.1.2 Advanced: Registration Options ................................................................................ 36
4.1.3 Call Settings ............................................................................................................... 36
4.1.4 Media Settings .......................................................................................................... 37
4.2 FXS Batch Binding SIP ........................................................................................................... 37
4.3 Batch Create SIP ................................................................................................................... 38
4.4 Advanced SIP Settings .......................................................................................................... 38
4.4.1 Networking ................................................................................................................ 38
4.4.2 Parsing and Compatibility ......................................................................................... 40
4.4.3 Security ...................................................................................................................... 41
4.4.4 Media ........................................................................................................................ 42
4.5 Sip Account Security ............................................................................................................. 43
5. Routing ........................................................................................................................................... 45
5.1 Call Routing Rules ................................................................................................................. 45
5.2 Groups .................................................................................................................................. 48
5.3 Batch Create Rules ............................................................................................................... 49
5.4 Advanced settings ................................................................................................................ 50
6. Network .......................................................................................................................................... 51
6.1 Network Settings .................................................................................................................. 51
6.2 VPN Settings ......................................................................................................................... 53
6.3 DDNS Settings....................................................................................................................... 53
6.4 Toolkit ................................................................................................................................... 54
6.5 Firewall Settings ................................................................................................................... 55
6.6 Security Rules ....................................................................................................................... 56
7. Advanced ........................................................................................................................................ 58
7.1 Asterisk API ........................................................................................................................... 58
7.2 Asterisk CLI ........................................................................................................................... 60
7.3 Asterisk File Editor ................................................................................................................ 61
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8. Logs ................................................................................................................................................ 62
8.1 Log Settings .......................................................................................................................... 62
8.2 CDR ....................................................................................................................................... 65
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1. Overview
1.1 What is Analog Gateway Series?
OpenVox Analog Gateway Series 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).
The Modular Design Analog Gateway Series 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 Series 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.
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1.2 Sample Application
Figure 1-2-1 Topological Graph
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1.3 Product Appearance
The picture below is appearance of Analog Series Gateway.
Figure 1-3-1 Product Appearance
Figure 1-3-2 Front Panel
1: Reset button
2: Power Indicator
3: System LED
4: Analog Telephone Interface (2)
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5: Channels State Indicator
1.4 Main Features
➢ Modular design
➢ Based on Asterisk○R
➢ Editable Asterisk
○R
configuration file
➢ 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
1.5 Physical Information
Table 1-5-1 Description of Physical Information
Overall Unit
Product Name
VS-GW1600(V2)
VS-GW2120(V2)
Interface
FXS/FXO Port
16/24/32/40
48/56/64/72/80/88
Lan
2
USB
1 (Supports USB storage)
TF
1 (Supports TF Card storage)
Console
1
Specifications
Weight
3310g(without module
board)
5710g(without module
board)
Dimension
44cm*4.5cm*30cm
48.3cm*33.1cm*8.8cm
Maximum Power Consumption
60W
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Power Supply
100-240V/1-2A
Operaton Temperature
0℃ ~ 40℃
Operaton Humidity Range
10% ~ 90%
Storage Temperature Range
-20℃ ~ 70℃
Module Board
Product Name
VS-GWM820-O
VS-GWM820-S
VS-GWM820-O/S
VS-GWM820 Control
Board
Interface
FXS/FXO Port
8 FXO
8 FXS
4 FXO,4 FXS
-
Lan - 2
USB
-
1 (Supports USB
storage)
TF
-
1 (Supports TF Card
storage)
Console - 1
Specifications
Weight
150g
140g
145g
190g
Dimension
13cm*2.1cm*20cm
Maximum Power
Consumption
3W
12W
6W
7W
Operaton
Temperature
0℃ ~ 40℃
Operaton
Humidity Range
10% ~ 90%
Storage
Temperature
Range
-20℃ ~ 70℃
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1.6 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. ETH1
only can enter the ip of slot 1, and ETH2 can access every slot. You can choose either of them, that
depends on your demand.
Figure 1-6-1 Login Interface
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2. System
2.1 Status
On the “Status” page, you will see Port/SIP/Routing/Network information and status.
Figure 2-1-1 System Status
2.2 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
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as your city.
POSIX TZ String
Posix time zone strings.
NTP Server 1
Time server domain or hostname. For example, [time.asia.apple.com].
NTP Server 2
The first reserved NTP server. For example, [time.windows.com].
NTP Server 3
The second reserved NTP server. For example, [time.nist.gov].
Auto-Sync from
NTP
Whether enable automatically synchronize from NTP server or not. ON is
enable, OFF is disable this function.
Sync from NTP
Sync time from NTP server.
Sync from Client
Sync time from local machine.
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.
2.3 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
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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
Define your username and password to manage your gateway,
without space here. Allowed characters
"-_+. < >&0-9a-zA-Z". Length: 1-32 characters.
Password
Allowed characters "-_+. < >&0-9a-zA-Z". Length: 4-32 characters.
Confirm Password
Please input the same password as 'Password' above.
Login Mode
Select the mode of login.
HTTP Port
Specify the web server port number.
HTTPS Port
Specify the web server port number.
Port
SSH login port number.
Figure 2-3-1 Login Settings
Notice: Whenever you do some changes, do not forget to save your configuration.
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2.4 General
2.4.1 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
2.4.2 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 help system work more efficient.
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2.4.3 SNMP Agent
Figure 2-4-3 SNMP Agent
2.5 Tools
On the “Tools” pages, there are reboot, update, upload, backup and restore toolkits.
You can choose system reboot and Asterisk reboot separately.
Figure 2-5-1 Reboot Prompt
If you press “Yes”, 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.
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 2-5-2 Update Firmware
If you want to store your previous configuration, you can first backup configuration, then you can
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upload configuration directly. That will be very convenient for you. Notice, the version of backup
and current firmware should be same, otherwise, it would not take effect.
Figure 2-5-3 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 2-5-4 Factory Reset
2.6 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 2-6-1 System Information
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2.7 Config Record
Figure 2-7-1 Config Record Interface
Turn on the configuration record function to roll back by operation record.
2.8 Config Label
Figure 2-8-1 Config Label Interface
Create a new label and make a manual backup to roll back.
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3. Analog
You can see much information about your ports on this page.
3.1 Channel Settings
Figure 3-1-1 Channel System
On this page, you can see every port status, and click action button to configure the port.
Figure 3-1-2 FXO Port Configure
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Figure 3-1-3 FXS Port Configure
3.2 Pickup Settings
Call pickup is a feature used in a telephone system that allows one to answer someone else's
telephone call. You can set the “Time Out” and “Number” parameters either globally or separately
for each port. The feature is accessed by pressing a special sequence of numbers which you set as
“Number” parameter on the telephone set when it is enabled this function.
Figure 3-2-1 Pickup Configure
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Table 3-2-1 Definition of Pickup
Options
Definition
Enable
ON (enabled), OFF (disabled)
Time Out
Set the timeout, in milliseconds (ms). Note: You can only enter numbers.
Number
Pickup number
3.3 Dial Matching Table
Dialing rules is used to effectively judge whether the received number sequence is complete, in
order to timely end receiving number and send out number
The correct use of dial-up rules, helps to shorten the turn-on time of phone call
Figure 3-3-1 Port Configure
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3.4 Advanced Settings
Figure 3-4-1 General Configuration
Table 3-4-1 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.
Max flash time
Max flash time.(in milliseconds).
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“#”as Ending Dial Key
Turn on/off Ending Dial Key.
Checking SIP Status
Turn on/off SIP Account registration status checking.
Figure 3-4-2 Caller ID
Table 3-4-2 Instruction of Caller ID
Options
Definition
The pattern of sending
CID
Some countries (UK) have ring tones with different ring
tones(ring-ring), which means the caller ID needs to be set later on,
and not just after the first ring, as per the default (1).
Waiting time before
sending CID
How long we will waiting before sending the CID on the channel. (in
milliseconds).
Sending polarity
reversal (DTMF Only)
Send polarity reversal before sending the CID on the channel.
Start code (DTMF Only)
Start code.
Stop code (DTMF Only)
Stop code.
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Figure 3-4-3 Hardware Gain
Table 3-4-3 Instruction of Hardware gain
Options
Definition
FXS Rx gain
Set the FXS port Rx gain. Range: from -150 to 120. Select -35, 0 or 35.
FXS Tx gain
Set the FXS port Tx gain. Range: from -150 to 120. Select -35, 0 or 35.
Figure 3-4-4 Fax Configuration
Table 3-4-4 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.
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Figure 3-4-5 Country Configuration
Table 3-4-5 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.)
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3.5 Special Function Keys
Figure 3-5-1 Function keys
Table 3-5-1 Definition of Function keys
Options
Definition
None Keys Blind Transfer
Turn on the switch: A calls B, A press *38(can be customized) and
calls C phone number, C picks up and talk with A
Turn off the switch: A calls B, A press *38(can be customized) and
calls C phone number, A hangs up, C picks up and talk with B
Blind Transfer
When turn on the Blind Transfer, configure the dialplan before
transferring the call (*38 by default)
Asked Transfer
When turn on the Asked Transfer, configure the dialplan before
transferring the call (*38 by default)
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3.6 FXO
Figure 3-6-1 FXO
Table 3-6-1 Instruction of FXO
Options
Definition
Busy Detect
Busy Detection is used for detecting busy signal or far end hang-up
Busy Count
Configure the number of busy tones the user will hear before hanging up the
call when Busy Detect is enabled.
Busy country
Select country for tone settings.
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3.7 FXS
Figure 3-7-1 FXS
Table 3-7-1 Instruction of FXS
Options
Definition
The pattern of
sending CID
Some countries (UK) have ring tones with different ring tones(ring-ring),
which means the caller ID needs to be set later on, and not just after the
first ring, as per the default (1).
Waiting time before
sending CID
How long we will be waiting before sending the CID on the channel. (in
milliseconds).
Sending polarity
reversal (DTMF Only)
Send polarity reversal before sending the CID on the channel.
Start code (DTMF
Only)
Start code.
Stop code (DTMF
Only)
Stop code.
Flash/Wink
Turn on/off Flash/wink.
Min flash time
Min flash time. (in milliseconds).
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Max flash time
Max flash time. (in milliseconds).
“#”as Ending Dial Key
Turn on/off Ending Dial Key.
Offhook-antishake
The anti-jitter delay value when the gateway FXS port detects the
off-hook signal. The setting value is from 8ms to 2048ms (multiple of 8)
and the default value is 32ms.
3.8 Driver
Figure 3-8-1 Driver
Table 3-8-1 Instruction of Driver
Options
Definition
Codec
Set the global encoding: ulaw, alaw.
Impedance
Configuration for impedance.
cidbeforering
Swith to handle irregular CID function
Cidbuflen
CID media stream length byte size
Cutcidbufheadlen
CID media stream header length byte size
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Fixedtimepolarity
Transmit polarity line reversal signal delay time
FXO Rx gain
Set FXO to terminal gain. Range: from -150 to 120.
FXO Tx gain
Set FXO to IP gain. Range: from -150 to 120.
FXS Rx gain
Set FXS to terminal gain. Select -35, 0 or 35.
FXS Tx gain
Set FXS to IP gain. Select -35, 0 or 35.
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4. SIP
4.1 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.
4.1.1 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.)
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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
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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
None---Not registering;
Endpoint registers with this gateway---When register as this type, it means the GSM
gateway acts as a SIP server, and SIP endpoints register to the gateway;
This gateway registers with the endpoint---When register as this type, it means the
GSM gateway acts as a client, and the endpoint should be register to a SIP server;
Hostname or IP
Address
IP address or hostname of the endpoint or 'dynamic' if the endpoint has a
dynamic IP address. This will require registration.
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.
No---Use Rport if the remote side says to use it.
Force Rport on---Force Rport to always be on.
Yes---Force Rport to always be on and perform comedia RTP
handling.
Rport if requested and comedia---Use Rport if the remote side
says to use it and perform comedia RTP handling.
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4.1.2 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.
Custom Registery
Custom Registery On / Off.
Enable Outboundproxy
to Host
Outboundproxy to Host On / Off.
4.1.3 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).
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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.
4.1.4 Media Settings
Ta ble 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.
4.2 FXS Batch Binding SIP
If you want binding batch Sip accounts to FXS port, you can configure this page. Look out: this is
only used when “This gateway registers with the endpoint” work mode.
Figure 4-2-1 FXS Batch Binding SIP
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4.3 Batch Create SIP
If you want add batch Sip accounts, you can configure this page. You can choose all the register
mode.
Figure 4-3-1 Batch SIP Endpoints
4.4 Advanced SIP Settings
4.4.1 Networking
Table 4-4-1 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
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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.
4.4.1.1 NAT Settings
Table 4-4-2 Definition 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 of allowing
your users to register at the same address as a SIP provider.
Externally Mapped
TCP Port
The externally mapped TCP port, when the gateway is behind a static NAT
or PAT
External Address
The external address (and optional TCP port) of the NAT.
External Address = hostname[:port] specifies a static address[:port] to be
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used in SIP and SDP 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.
4.4.1.2 RTP Settings
Table 4-4-3 Definition of NAT 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
4.4.2 Parsing and Compatibility
Table 4-4-4 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
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SDP Owner
Allows you to change the username filed in the SDP owner string.
This filed MUST NOT contain spaces.
Disallowed SIP
Methods
The external hostname (and optional TCP port) of the NAT.
Shrink Caller ID
The shrinkcallerid function removes '(', ' ', ')', non-trailing '.', and '-' not in
square brackets. For example, the caller id value 555.5555 becomes
5555555 when this option is enabled. Disabling this option results in no
modification of the caller id value, which is necessary when the caller id
represents something that must be preserved. By default this option is on.
Maximum
Registration Expiry
Maximum allowed time of incoming registrations and subscriptions
(seconds).
Minimum
Registration Expiry
Minimum length of registrations/subscriptions (default 60).
Default Registration
Expiry
Default length of incoming/outgoing registration.
Registration
Timeout
How often, in seconds, to retry registration calls. Default 20 seconds.
Number of
Registration
Attempts Enter '0'
for unlimited
Number of registration attempts before we give up. 0 = continue forever,
hammering the other server until it accepts the registration. Default is 0
tries, continue forever.
4.4.3 Security
Table 4-4-5 Instruction of Security
Options
Definition
Match Auth Username
If available, match user entry using the 'username' field from the
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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.
4.4.4 Media
Table 4-4-6 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
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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
4.5 Sip Account Security
This analog gateway support TLS protocl for encrypting calls. On the one hand, it can worked as TLS
server, generate the session keys used for the secure connection. On the other hand, it also can be
registered as a client, upload the key files provied by the server.
Figure 4-5-1 TLS settings
Table 4-5-1 Instruction of TLS
Options
Definition
TLS Enable
Enable or disable DTLS-SRTP support.
TLS Verify Server
Enable or disable tls verify server(default is no).
Port
Specify the port for remote connection.
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TLS Client Method
Values include tlsv1, sslv3, sslv2, Specify protocol for outbound client
connections, default is sslv2.
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5. Routing
The gateway embraces the flexible and friendly routing settings for user. It supports up to 512
routing rules and about 100 pairs of calleeID/callerID manipulations can be set in a rule. It support
DID function The gateway support trunk group and trunk priority management.
5.1 Call Routing Rules
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.
will show current routing rules. Otherwise you can set up unlimited routing rules.
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
China Telecom. Called transform adds 086 as prefix, and Change the last two number to 88.
Figure 5-1-1
processing rules
prepend
prefix
Match pattern
SdfR
StA
RdfR
Caller Name
Calling Transformation
086
159
xxxxxxxx
4
0755
China
telecom
Called transformation
086
136
xxxxxxx
2
88 N/A
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You can click button to set up your routings.
Figure 5-1-2 Example of Setup 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 5-1-2 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
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Table 5-1-3 Definition of Advance Routing Rule
Options
Definition
CalleeID/callerID
Manipulation
A Dial Pattern is a unique set of digits that will select this route and send the
call to the designated trunks. If a dialed pattern matches this route, no
subsequent routes will be tried. If Time Groups are enabled, subsequent
routes will be checked for matches outside of the designated time(s).
X matches any digit from 0-9
Z matches any digit from 1-9
N matches any digit from 2-9
[1237-9]matches any digit in the brackets (example: 1,2,3,7,8,9)
. 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.
Mach 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): The amount of digits to be resevered from
the right end of the number. If the value of this item under the length of the
current number, the whole number will be reserverd.
StA(Suffix to Add): Designated information to be added to the right end of
the current number.
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Caller Name: What caller name would you like to set before sending this call
to the endpoint.
Disabled Caller Number Change : Disable the caller number change, and fixed
caller number match pattern.
Time Patterns that
will use this Route
Time Patterns that will use this Route help
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.
5.2 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.
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.
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Figure 5-2-2 Create a Group
Figure 5-2-3 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’).
5.3 Batch Create Rules
If you bind telephone for each FXO port and want to establish separate call routings for them. For
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convenience, you can batch create call routing rules for each FXO port at once in this page.
Figure 5-3-1 Batch Create Rules
5.4 Advanced settings
In the default mode, FXS and SIP are bound one-to-one. After the FXS port enables flexible routing,
it is compatible with the original sip extension and no longer required to bind SIP accounts. You can
set related routing rules. In this case, you can choose to enable the internal FXS/SIP call function.
Figure 5-4-1 Advanced settings
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6. Network
On “Network” page, there are “Network Settings”, “VPN Settings”, “DDNS Settings”, and “Toolkit”.
6.1 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.
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Type
The method to get IP.
Factory: Getting IP address by Slot Number (System → information to check
slot number).
Static: manually set up your gateway IP.
DHCP: automatically get IP from your local LAN.
MAC
Physical address of your network interface.
Address
The IP address of your gateway.
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
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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.
6.2 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 6-2-1 VPN Interface
6.3 DDNS Settings
You can enable or disable DDNS (dynamic domain name server).
Figure 6-3-1 DDNS Interface
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Table 6-3-1 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.
6.4 Toolkit
It is used to check network connectivity. Support Ping command on web GUI.
Figure 6-4-1 Network Connectivity Checking
Figure 6-4-2 Channel Recording
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Figure 6-4-3 Capture Network Data
Table 6-4-1 Definition of Channel Recording
Options
Definition
Interface
The name of network interface.
Source host
Capture the data of source host you specified
Destination host
Capture the data of destination host you specified
Port
Capture the data of port you specified
Channel
Capture the data of channel you specified
Tcpdump Option
Parameter
The tool of tcpdump capture network data by parameter option specified.
6.5 Firewall Settings
Figure 7-6-1 Firewall Settings
Table 7-6-1 Definition of Firewall Settings
Options
Definition
Firewall Enable
If you want to use White/Black List, and security rules, you must enable this
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option.
Ping Enable
Whether to enable the Ping function. If the status is OFF: disable ping, the
gateway does not allow ping.
Figure 7-6-2 White/Black List Settings
Table 7-6-2 Definition of White/Black List Settings
Options
Definition
White/Black List Enable
To enable White/Black list or not.
List IP
IP is separated only by "," character.
6.6 Security Rules
Figure 7-7-1 Security Rules Settings
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Click "submit" button to submit and apply configuration.
If "List IP Settings" has no problem, you will see popup window like below. Please read the warning
and tips carefully. And Click "Apply" button in 1 minute. If time runs out, this window will close
automatically.
Figure 7-7-2 Security Rules Apply
If you see the windows like below. It means your configuration has been applied successfully.
Figure 7-7-3 Security Rules Apply
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7. Advanced
7.1 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 separato
r. 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.
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0
Permit
If you want to permit many hosts or network, use char & as separat
or.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 172.16.80.16/255.255.0.0 is allowed to access the
gateway API. Please refer to the following figure to access the gateway API by putty. 172.16.80.16
is the gateway’s IP, and 5038 is its API port.
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Figure 7-1-2 Putty Access
7.2 Asterisk CLI
In this page, you are allowed to run Asterisk commands.
Figure 7-2-1 Asterisk Command Interface
Table 7-2-1 Definition of Asterisk API
Options
Definition
Command
Type your Asterisk CLI commands here to check or debug your gateway.
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If you type “help” or “?” and execute it, the page will show you the executable commands.
7.3 Asterisk File Editor
On this page, you are allowed to edit and create configuration files.
Click the file to edit.
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
8.1 Log Settings
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 are unavailable. And the same with other log pages.
Figure 8-1-1 Logs Settings
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Table 8-1-1 Definition of LOG
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 each channel.
Auto clean:
(Call Detail
Record)
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.
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Figure 8-1-2 System Logs Output
Notice: The same to Asterisk Logs, SIP Logs and DAHDI Logs.
With Syslog software, you can monitor and store the gateway's logs and CDRs locally on the PC.
Figure 8-1-3 Syslog
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8.2 CDR
You can scan every call detail record in this page. We also provide the filter for you to search some
specific records.
Figure 8-2-1 Call Detail Record
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