Analog Gateway User Manual
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OpenVox Communication Co Ltd
Analog Gateway User Manual
Version 1.0
Analog Gateway User Manual
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OpenVox Communication Co. ,LTD. URL: www.openvox.cn
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.
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
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Contents
1. Overview ................................................................................................................................................................ 8
What is Analog Gateway Series?........................................................................................................................ 8
Sample Application ............................................................................................................................................ 8
Product Appearance .......................................................................................................................................... 9
Main Features .................................................................................................................................................... 9
Physical Information ........................................................................................................................................ 10
Software ........................................................................................................................................................... 10
2. System .................................................................................................................................................................. 12
Status ............................................................................................................................................................... 12
Time ................................................................................................................................................................. 12
Login Settings ................................................................................................................................................... 13
General ............................................................................................................................................................ 15
Language Settings .................................................................................................................................... 15
Scheduled Reboot .................................................................................................................................... 15
Tools ................................................................................................................................................................. 15
Information ...................................................................................................................................................... 17
3. Analog .................................................................................................................................................................. 18
Channel Settings .............................................................................................................................................. 18
Pickup Settings ................................................................................................................................................. 19
Dial Matching Table ......................................................................................................................................... 20
Advanced Settings............................................................................................................................................ 20
Special Function Keys ....................................................................................................................................... 24
4. SIP ........................................................................................................................................................................ 25
SIP Endpoints ................................................................................................................................................... 25
Main Endpoint Settings ............................................................................................................................ 25
Advanced: Registration Options .............................................................................................................. 27
Call Settings .............................................................................................................................................. 28
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Advanced: Signaling Settings ................................................................................................................... 29
Advanced: Timer Settings ........................................................................................................................ 30
Media Settings ......................................................................................................................................... 31
FXS Batch Binding SIP ....................................................................................................................................... 31
Batch Create SIP ............................................................................................................................................... 32
Advanced SIP Settings ...................................................................................................................................... 32
Networking .............................................................................................................................................. 32
NAT Settings ............................................................................................................................................. 33
RTP Settings ............................................................................................................................................. 34
Parsing and Compatibility ........................................................................................................................ 34
Security .................................................................................................................................................... 35
Media ....................................................................................................................................................... 36
Sip Account Security ........................................................................................................................................ 37
5. Routing ................................................................................................................................................................. 38
Call Routing Rules ............................................................................................................................................ 38
Groups ............................................................................................................................................................. 41
Batch Create Rules ........................................................................................................................................... 42
6. Network ............................................................................................................................................................... 44
Network Settings ............................................................................................................................................. 44
VPN Settings ............................................................................................................................................. 46
DDNS Settings .......................................................................................................................................... 46
Toolkit ...................................................................................................................................................... 47
7. Advanced ............................................................................................................................................................. 49
Asterisk API ...................................................................................................................................................... 49
Asterisk CLI ....................................................................................................................................................... 51
Asterisk File Editor ........................................................................................................................................... 52
8. Logs ...................................................................................................................................................................... 53
Log Settings ...................................................................................................................................................... 53
System .............................................................................................................................................................. 55
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CDR .................................................................................................................................................................. 55
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1. Overview
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.
Sample Application
Figure 1-2-1 Topological Graph
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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)
5: Channels State Indicator
Main Features
Modular design
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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
Physical Information
Table 1-5-1 Description of Physical Information
Weight
200g
Size
22cm*2cm*12cm
Temperature
-40~125°C (Storage)
0~50°C (Operation)
Operation humidity
10%~90% non-condensing
Power source
12V DC/2A
Max power
11W
Software
Default IP: 172.16.99.1
Username: admin
Password: admin
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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
Status
On the “Status” page, you will see Port/SIP/Routing/Network information and status.
Figure 2-1-1 System Status
Time
Table 2-2-1 Description of Time Settings
Options
Definition
System Time
Your gateway system time.
Time Zone
The world time zone. Please select the one which is the same or the
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closest as your city.
POSIX TZ String
Posix time zone strings.
NTP Server 1
Time server domain or hostname. For example, [time.asia.apple.com].
NTP Server 2
The first reserved NTP server. For example, [time.windows.com].
NTP Server 3
The second reserved NTP server. For example, [time.nist.gov].
Auto-Sync from NTP
Whether enable automatically synchronize from NTP server or not. ON
is enable, OFF is disable this function.
Sync from NTP
Sync time from NTP server.
Sync from Client
Sync time from local machine.
For example, you can configure like this:
Figure 2-2-1 Time Settings
You can set your gateway time Sync from NTP or Sync from Client by pressing different buttons.
Login Settings
Your gateway doesn't have administration role. All you can do here is to reset what new username and password
to manage your gateway. And it has all privileges to operate your gateway. You can modify both your “Web Login
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Settings” and “SSH Login Settings”. If you have changed these settings, you don’t need to log out, just rewriting
your new user name and password will be OK.
Table 2-3-1 Description of Login Settings
Options
Definition
User Name
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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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”, those will be ok.
Figure 2-4-1 Language Settings
Scheduled Reboot
If switch it on, you can manage your gateway to reboot automatically as you like. There are four reboot types for
you to choose, “By Day, By Week, By Month and By Running Time”.
Figure 2-4-2 Reboot Types
If use your system frequently, you can set this enable, it can helps system work more efficient.
Tools
On the “Tools” pages, there are reboot, update, upload, backup and restore toolkits.
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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 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
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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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3. Analog
You can see much information about your ports on this page.
Channel Settings
Figure 3-1-1 Channel System
On this page, you can see every port status, and click action button to configure the port.
Figure 3-1-2 FXO Port Configure
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Figure 3-1-3 FXS Port Configure
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
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
Advanced Settings
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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).
“#”as Ending Dial Key
Turn on/off Ending Dial Key.
Checking SIP Status
Turn on/off SIP Account registration status checking.
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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.
Figure 3-4-3 Hardware Gain
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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.
Figure 3-4-5 Country Configuration
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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.)
Special Function Keys
Figure 3-5-1 Function keys
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4. SIP
SIP Endpoints
This page shows everything about your SIP, you can see status of each SIP.
Figure 4-1-1 SIP Status
You can click button to add a new SIP endpoint, and if you want to modify existed
endpoints, you can click button.
Main Endpoint Settings
There are 3 kinds of registration types for choose. You can choose “Anonymous, Endpoint registers with this
gateway or This gateway registers with the endpoint”.
You can configure as follows:
If you set up a SIP endpoint by registration “None” to a server, then you can’t register other SIP endpoints to this
server. (If you add other SIP endpoints, this will cause Out-band Routes and Trunks confused.)
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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.
Advanced: Registration Options
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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.
Call Settings
Table 4-1-3 Definition of Call Options
Options
Definition
DTMF Mode
Set default DTMF Mode for sending DTMF. Default: rfc2833.
Other options: 'info', SIP INFO message (application/dtmf-relay);
'Inband', Inband audio (require 64kbit codec -alaw, ulaw).
Call Limit
Setting a call-limit will cause calls above the limit not to be accepted.
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Trust
Remote-Party-ID
Whether or not the Remote-Party-ID header should be trusted.
Send
Remote-Party-ID
Whether or not to send the Remote-Party-ID header.
Remote Party ID
Format
How to set the Remote-Party-ID header: from Remote-Party-ID or
from P-Asserted-Identity.
Caller ID Presentation
Whether or not to display Caller ID.
Advanced: Signaling Settings
Table 4-1-4 Definition of Signaling Options
Options
Definition
Progress Inband
If we should generate in-band ringing.
Always use ‘never’ to never use in-band signaling, even in cases
where some buggy devices might not render it.
Valid values: yes, no never.
Default: never.
Allow
Overlap
Dialing
Allow Overlap Dialing: Whether or not to allow overlap dialing.
Disabled by default.
Append
user=phone to
URI
Whether or not to add ‘; user=phone’ to URIs that contain a valid phone
number.
Add Q.850
Reason Headers
Whether or not to add Reason header and to use it if it is available.
Honor SDP Version
By default, the gateway will honor the session version number in SDP
packets and will only modify the SDP session if the version number
change. Turn this option off to force the gateway to ignore the SDP
session version number and treat all SDP data as new data. This is
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required for devices that send non-standard SDP packets (observed
with Microsoft
OCS). By default this option is on.
Allow Transfers
Whether or not to globally enable transfers. Choosing 'no' will disable all
transfers (unless enabled in peers or users). Default is enabled.
Allow
Promiscuous
Redirects
Whether or not to allow 302 or REDIR to non-local SIP address.
Note that promiscredir when redirects are made to the local system will
cause loops since this gateway is incapable of performing a "hairpin" call.
Max Forwards
Setting for the SIP Max-Forwards header (loop prevention).
Send TRYING
on REGISTER
Send a 100 Trying when the endpoint registers.
Advanced: Timer Settings
Table 4-1-5 Definition of Timer Options
Options
Definition
Default T1 Timer
This timer is used primarily in INVITE transactions. The default for Timer
T1 is 500ms or the measured run-trip time between the gateway and the
device if you have qualify=yes for the device.
Call Setup Timer
If a provisional response is not received in this amount of time, the call will
auto-congest. Defaults to 64 times the default T1 timer.
Session Timers
Session-Timers feature operates in the following three modes: originate,
Request and run session-timers always; accept, run session-timers only when
requested by other UA; refuse, do not run session timers in any case.
Minimum
Session Refresh
Interval
Minimum session refresh interval in seconds. Default is 90secs.
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Maximum
Session Refresh
Interval
Maximum session refresh interval in seconds. Defaults to 1800secs.
Session Refresher
The session refresher, uac or uas. Defaults to uas.
Media Settings
Table 4-1-6 Definition of Media Settings
Options
Definition
Media Settings
Select codec from the drop down list. Codecs should be different for each
Codec Priority.
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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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
Advanced SIP Settings
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 users on the
Internet specifying a port in a SIP peer definition or when dialing
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outbound calls with suppress SRV lookups for that peer or call.
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 used in
SIP and SDP messages.Examples:
External Address = 12.34.56.78
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External Address = 12.34.56.78:9900
External
Hostname
The external hostname (and optional TCP port) of the NAT.
External Hostname = hostname[:port] is similar to External Address. Examples:
External Hostname = foo.dyndns.net
Hostname Refresh
Interval
How often to perform a hostname lookup. This can be useful when your NAT
device lets you choose the port mapping, but the IP address is dynamic. Beware,
you might suffer from service disruption when the name server resolution fails.
RTP Settings
Table 4-4-3 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
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
SDP Owner
Allows you to change the username filed in the SDP owner
string.
This filed MUST NOT contain spaces.
Disallowed SIP
The external hostname (and optional TCP port) of the NAT.
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Methods
Shrink Caller ID
The shrinkcallerid function removes '(', ' ', ')', non-trailing '.', and
'-' not in square brackets. For example, the caller id value
555.5555 becomes 5555555 when this option is enabled.
Disabling this option results in no modification of the caller id
value, which is necessary when the caller id represents
something that must be preserved. By default this option is on.
Maximum
Registration Expiry
Maximum allowed time of incoming registrations and
subscriptions (seconds).
Minimum
Registration Expiry
Minimum length of registrations/subscriptions (default 60).
Default Registration
Expiry
Default length of incoming/outgoing registration.
Registration
Timeout
How often, in seconds, to retry registration calls. Default 20
seconds.
Number of
Registration
Attempts Enter '0'
for unlimited
Number of registration attempts before we give up. 0 = continue
forever, hammering the other server until it accepts the
registration. Default is 0 tries, continue forever.
Security
Table 4-4-5 Instruction of Security
Options
Definition
Match Auth Username
If available, match user entry using the 'username' field from the
authentication line instead of the 'from' field.
Realm
Realm for digest authentication. Realms MUST be globally unique according
to RFC 3261. Set this to your host name or domain name.
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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 4-4-6 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
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()
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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
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.
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.
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.
Caller Name: What caller name would you like to set before sending this call to the
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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.
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’).
Batch Create Rules
If you bind telephone for each FXO port and want to establish separate call routings for them. For convenience,
you can batch create call routing rules for each FXO port at once in this page.
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Figure 5-3-1 Batch Create Rules
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6. Network
On “Network” page, there are “Network Settings”, “VPN Settings”, “DDNS Settings”, and “Toolkit”.
Network Settings
There are three types of LAN port IP, Factory, Static and DHCP. Factory is the default type, and it is 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
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Interface
The name of network interface.
Type
The method to get IP.
Factory: Getting IP address by Slot Number (System
information to check slot number).
Static: manually set up your gateway IP.
DHCP: automatically get IP from your local LAN.
MAC
Physical address of your network interface.
Address
The IP address of your gateway.
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.
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
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
Type
Set the type of DDNS server.
Username
Your DDNS account’s login name.
Password
Your DDNS account’s password.
Your domain
The domain to which your web server will belong.
Toolkit
It is used to check network connectivity. Support Ping command on web GUI.
Figure 6-4-1 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.
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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 192.168.1.0/255.2
55.255.0&10.0.0.0/255.0.0.0
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Permit
If you want to permit many hosts or network, use char
& as separator.Example: 0.0.0.0/0.0.0.0 or 192.168.1.0/255.
255.255.0&10.0.0.0/255.0.0.0
System
General information about the system and ability to run system
management commands, such as Shutdown, Restart, and
Reload.
Call
Information about channels and ability to set information in a
running channel.
Log
Logging information. Read-only. (Defined but not yet used.)
Verbose
Verbose information. Read-only. (Defined but not yet used.)
Command
Permission to run CLI commands. Write-only.
Agent
Information about queues and agents and ability to add queue
members to a queue.
User
Permission to send and receive UserEvent.
Config
Ability to read and write configuration files.
DTMF
Receive DTMF events. Read-only.
Reporting
Ability to get information about the system.
CDR
Output of cdr, manager, if loaded. Read-only.
Dialplan
Receive NewExten and Varset events. Read-only.
Originate
Permission to originate new calls. Write-only.
All
Select all or deselect all.
Once you set like the above figure, the host 172.16.80.16/255.255.0.0 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
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.
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
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
is 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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System
Figure 8-2-1 System Logs Output
Notice: The same to Asterisk Logs and SIP Logs.
CDR
You can scan every call detail records in this page. We also provide the filter for you to search some specific
records.
Figure 8-3-1 Call Detail Record
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