VS-GW1600-40S User Manual
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VS-GW1600-40S User Manual
OpenVox Communication Co.Ltd
Version1.0 (2014-01-22)
Full text
The overall layout adjustment
Version1.1(2014-04-15)
Full text
VS-GW1600-40S User Manual
OpenVox Communication Co.Ltd
Copyright
Copyright© 2013 OpenVox Inc. All rights reserved. No part of this document may be reproduced without
prior written permission.
Confidentiality
Information contained herein is of a highly sensitive nature and is confidential and proprietary to OpenVox
Inc. No part may be distributed, reproduced or disclosed orally or in written form to any party other than
the direct recipients without the express written consent of OpenVox Inc.
Disclaimer
OpenVox Inc. reserves the right to modify the design, characteristics, and products at any time without
notification or obligation and shall not be held liable for any error or damage of any kind resulting from the
use of this document.
OpenVox has made every effort to ensure that the information contained in this document is accurate and
complete; however, the contents of this document are subject to revision without notice. Please contact
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Trademarks
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Table of Contents
1. Overview ................................................................................................................................ 1
What is VS-GW1600-40S? 1
Sample Application 2
Product Appearance 2
Main Features 3
Physical Information 3
Software 3
2. System ................................................................................................................................... 4
Status 4
Time 5
Login Settings 6
General, Tools and Information 7
Language Settings 7
Scheduled Reboot 7
Reboot Tools 7
Information 9
3. Analog .................................................................................................................................... 9
Channel Settings 9
Dial Matching Table 10
Global Settings 11
4. SIP ........................................................................................................................................ 15
SIP Endpoints 15
Main Endpoint Settings 15
Advanced: Registration Options 18
Call Settings 19
Advanced: Signaling Settings 19
Advanced: Timer Settings 20
Media Settings 21
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Batch SIP Endpoint 21
Advanced SIP Settings 22
Networking 22
NAT Settings 22
Advanced: NAT Settings 23
Parsing and Compatibility 23
Security 24
Media 25
5. Network, Advanced and Logs ................................................................................................ 26
Network 26
Network Settings 26
OpenVPN Settings 29
DDNS Settings 29
Toolkit 30
Advanced 30
Asterisk API 30
Asterisk CLI 32
Asterisk File Editor 33
Logs 34
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1. Overview
What is VS-GW1600-40S?
OpenVox VoxStack Series Analog Gateway is an open source asterisk-based Analog VoIP Gateway
solution for SMBs and SOHOs. With friendly GUI and unique modular design, users may easily
setup their customized Gateway. Also secondary development can be completed through AMI
(Asterisk Management Interface).
There are three models with VoxStack series Analog Gateway, the VS-GW1202-8S,
VS-GW1202-16S and VS-GW1600-40S. There are 8 ports in VS-GW1202-8S. The Modular Design
Analog Gateways are ranging from 8 up to 40 ports, developed for interconnecting the PSTN
networks with a wide selection of codecs and signaling protocol, including G.711A, G.711U, G.729,
G.722, G.723, ILBC and GSM to quickly reduce communication expenses and maximize
cost-savings. With the unique design of the VoxStack gateway, it can support hot-swap. Users can
simply add or remove the modules for hardware expansion or exchange.
The VoxStack gateway designs with 2 LAN switch boards to provide stack ability on the hardware
upgrade. You can choose either of them.
The Analog gateway will be 100% compatible with Asterisk, Elastix, trixbox, 3CX, FreeSWITCH SIP
server and VOS VoIP operating platform.
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Sample Application
Figure 1-2-1 Topological Graph
Elastix Server
172.16.1.194
SIP Phone
Soft SIP Phone
VS-GW1600-40S
Internet
Analog Phone
Analog Cards
Product Appearance
The picture below is appearance of Analog Series Gateway.
Figure 1-3-1 Product Appearance
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Main Features
Modular and VoxStack 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
Physical Information
Table 1-5-1 Description of Physical Information
Weight
732g
Size
15cm*19cm*4.5cm
Temperature
-20~70°C (Storage)
0~40°C (Operation)
Operation humidity
10%~90% non-condensing
Power source
12V DC/4A
Max power
16W
LAN port
2
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. You can
choose either of them and they are the same.
Figure 1-6-1 LOGIN Interface
2. System
Status
On the “Status” page, you will see Port/SIP/Network information and status.
Figure 2-1-1 System Status
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Time
Table 2-2-1 Description of Time Settings
Options
Definition
System Time
Your gateway system time.
Time Zone
The world time zone. Please select the one which is the same or
the closest as your city.
POSIX TZ String
Posix time zone strings.
NTP Server 1
Time server domain or hostname. For example, [time.asia.apple.com].
NTP Server 2
The first reserved NTP server. For example, [time.windows.com].
NTP Server 3
The second reserved NTP server. For example, [time.nist.gov].
Auto-Sync from NTP
Whether enable automatically synchronize from NTP server or not. ON is
enable, OFF is disable this function.
Sync from NTP
Sync time from NTP server.
Sync from Client
Sync time from local machine.
For example, you can configure like this:
Figure 2-2-1 Time Settings
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You can set your gateway time Sync from NTP or Sync from Client by pressing different buttons.
Login Settings
Your gateway doesn't have administration role. All you can do here is to reset what new username
and password to manage your gateway. And it has all privileges to operate your gateway. You can
modify both your “Web Login Settings” and “SSH Login Settings”. If you have changed these
settings, you don’t need to log out, just rewriting your new user name and password will be OK.
Table 2-3-1 Description of Login Settings
Options
Definition
User Name
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.
Figure 2-3-1 Login Settings
Notice: Whenever you do some changes, do not forget to save your configuration.
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General, Tools and Information
Language Settings
You can choose different languages for your system. If you want to change language, you can
switch “Advanced” on, then “Download” your current language package. After that, you can
modify the package with the language you need. Then upload your modified packages, “Choose
File” and “Add”, those will be ok.
Figure 2-4-1 Language Settings
Scheduled Reboot
If switch it on, you can manage your gateway to reboot automatically as you like. There are four
reboot types for you to choose, “By Day, By Week, By Month and By Running Time”.
Figure 2-4-2 Reboot Types
If use your system frequently, you can set this enable, it can helps system work more efficient.
Reboot Tools
On the “Tools” pages, there are reboot, update, upload, backup and restore toolkits.
You can choose system reboot and Asterisk reboot separately.
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Figure 2-4-3 Reboot Prompt
If you press “Yes”, your system will reboot and all current calls will be dropped. Asterisk Reboot is
the same.
Table 2-4-1 Instruction of reboots
Options
Definition
System Reboot
This will turn off your gateway and then turn it back on. This
will drop all current calls.
Asterisk Reboot
This will restart Asterisk and drop all current calls.
We offer two kinds of update types for you, you can choose System Update or System Online
Update. System Online Update is an easier way to update your system.
Figure 2-4-4 Update Firmware
If you want to store your previous configuration, you can first backup configuration, then you can
upload configuration directly. That will be very convenient for you.
Figure 2-4-5 Upload and Backup
Sometimes there is something wrong with your gateway that you don’t know how to solve it,
mostly you will select factory reset. Then you just need to press a button, your gateway will be
reset to the factory status.
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Figure 2-4-6 Factory Reset
Information
On the “Information” page, there shows some basic information about the analog gateway. You
can see software and hardware version, storage usage, memory usage and some help information.
Figure 2-4-7 System Information
3. Analog
You can see much information about your ports on this page.
Channel Settings
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Figure 3-1-1 Channel System
On this page, you can see every port status, and click action button to configure the port.
Figure 3-1-2 Port Configure
Dial Matching Table
Dialing rules is used to effectively judge whether the received number sequence is complete, in
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order to timely end receiving number and send out number
The correct use of dial-up rules, helps to shorten the turn-on time of phone call
Figure 3-2-1 Port Configure
Global Settings
Figure 3-3-1 General Configuration
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Table 3-3-1 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.
Figure 3-3-2 Caller ID
Table 3-3-2 Instruction of Caller ID
Options
Definition
The pattern of
sending CID
Some countries(UK) have ring tones with different ring tones(ring-ring),
which means the caller ID needs to be set later on, and not just after the
first ring, as per the default(1).
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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 3-3-3 Hardware Gain
Table 3-3-3 Instruction of Hardware gain
Options
Definition
FXS Rx gain
Set the FXS port Rx gain. Range: -35, 0 or 35.
FXS Tx gain
Set the FXS port Tx gain. Range: -35, 0 or 35.
Figure 3-3-4 Fax Configuration
Table 3-3-4 Definition of Fax
Options
Definition
Mode
Set the transmission mode.
Rate
Set the rate of sending and receiving.
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Ecm
Enable/disable T.30 ECM (error correction mode) by default.
Figure 3-3-5 Country Configuration
Table 3-3-5 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.
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Info tone
Set of tones played with special information messages (e.g., number is
out of service.)
Stutter tone
4. SIP
SIP Endpoints
This page shows everything about your SIP, you can see status of each SIP.
Figure 4-1-1 SIP Status
You can click button to add a new SIP endpoint, and if you want to modify
existed endpoints, you can click button.
Main Endpoint Settings
There are 3 kinds of registration types for choose. You can choose “Anonymous, Endpoint registers
with this gateway or This gateway registers with the endpoint”.
You can configure as follows:
If you set up a SIP endpoint by registration “None” to a server, then you can’t register other SIP
endpoints to this server. (If you add other SIP endpoints, this will cause Out-band Routes and
Trunks confused.)
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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
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“None”, except name and password.
Figure 4-1-4 Register to Server
Table 4-1-1 Definition of SIP Options
Options
Definition
Name
A name which is able to read by human. And it’s only used for user’s
reference.
Username
User Name the endpoint will use to authenticate with the gateway.
Password
Password the endpoint will use to authenticate with the gateway. Allowed
characters.
Registration
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.
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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
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.
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Call Settings
Table 4-1-3 Definition of Call Options
Options
Definition
DTMF Mode
Set default DTMF Mode for sending DTMF. Default: rfc2833.
Other options: 'info', SIP INFO message (application/dtmf-relay);
'Inband', Inband audio (require 64kbit codec -alaw, ulaw).
Call Limit
Setting a call-limit will cause calls above the limit not to be accepted.
Trust
Remote-Party-ID
Whether or not the Remote-Party-ID header should be trusted.
Send
Remote-Party-ID
Whether or not to send the Remote-Party-ID header.
Remote Party ID
Format
How to set the Remote-Party-ID header: from Remote-Party-ID or from
P-Asserted-Identity.
Caller ID
Presentation
Whether or not to display Caller ID.
Advanced: Signaling Settings
Table 4-1-4 Definition of Signaling Options
Options
Definition
Progress Inband
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).
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.
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Honor SDP Version
By default, the gateway will honor the session version number in SDP
packets and will only modify the SDP session if the version number
change. Turn this option off to force the gateway to ignore the SDP session
version number and treat all SDP data as new data. This is required for
devices that send non-standard SDP packets (observed with Microsoft
OCS). By default this option is on.
Allow Transfers
Whether or not to globally enable transfers. Choosing 'no' will disable all
transfers (unless enabled in peers or users). Default is enabled.
Allow Promiscuous
Redirects
Whether or not to allow 302 or REDIR to non-local SIP address.
Note that promiscredir when redirects are made to the local system will
cause loops since this gateway is incapable of performing a "hairpin" call.
Max Forwards
Setting for the SIP Max-Forwards header (loop prevention).
Send TRYING on
REGISTER
Send a 100 Trying when the endpoint registers.
Outbound Proxy
A proxy to which the gateway will send all outbound signaling instead of
sending signaling directly to endpoints.
Advanced: Timer Settings
Table 4-1-5 Definition of Timer Options
Options
Definition
Default T1 Timer
This timer is used primarily in INVITE transactions. The default for Timer
T1 is 500ms or the measured run-trip time between the gateway and the
device if you have qualify=yes for the device.
Call Setup Timer
If a provisional response is not received in this amount of time, the call
will auto-congest. Defaults to 64 times the default T1 timer.
Session Timers
Session-Timers feature operates in the following three modes: originate,
Request and run session-timers always; accept, run session-timers only
when requested by other UA; refuse, do not run session timers in any
case.
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Minimum Session
Refresh Interval
Minimum session refresh interval in seconds. Default is 90secs.
Maximum Session
Refresh Interval
Maximum session refresh interval in seconds. Defaults to 1800secs.
Session Refresher
The session refresher, uac or uas. Defaults to uas.
Media Settings
Table 4-1-6 Definition of Media Settings
Options
Definition
Media Settings
Select codec from the drop down list. Codecs should be different
for each Codec Priority.
Batch SIP Endpoint
If you want add batch Sip accounts, you can configure this page. Look out: this is only used when
“This gateway registers with the endpoint” work mode.
Figure 4-2-1 Batch SIP Endpoint
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Advanced SIP Settings
Networking
Table 4-3-1 Definition of Networking Options
Options
Definition
UDP Bind Port
Choose a port on which to listen for UDP traffic.
Enable TCP
Enable server for incoming TCP connection (default is no).
TCP Bind Port
Choose a port on which to listen for TCP traffic.
TCP Authentication
Timeout
The maximum number of seconds a client has to authenticate. If the client
does not authenticate before this timeout expires, the client will be
disconnected.(default value is: 30 seconds).
TCP Authentication
Limit
The maximum number of unauthenticated sessions that will be
allowed to connect at any given time(default is:50).
Enable Hostname
Lookup
Enable DNS SRV lookups on outbound calls Note: the gateway only uses
the first host in SRV records Disabling DNS SRV lookups disables the ability
to place SIP calls based on domain names to some other SIP users on the
Internet specifying a port in a SIP peer definition or when dialing
outbound calls with suppress SRV lookups for that peer or call.
Enable Internal SIP
Call
Whether enable the internal SIP calls or not when you select the
registration option "Endpoint registers with this gateway".
Internal SIP Call
Prefix
Specify a prefix before routing the internal calls.
NAT Settings
Table 4-3-2 Definition of NAT Settings
Options
Definition
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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.
Advanced: NAT Settings
Table 4-3-3 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
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Table 4-3-4 Instruction of Parsing and Compatibility
Options
Definition
Strict RFC
Interpretation
Check header tags, character conversion in URIs, and multiline headers
for strict SIP compatibility(default is yes)
Send Compact
Headers
Send compact SIP headers
SDP Owner
Allows you to change the username filed in the SDP owner string.
This filed MUST NOT contain spaces.
Disallowed SIP
Methods
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.
Security
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Table 4-3-5 Instruction of Security
Options
Definition
Match Auth
Username
If available, match user entry using the 'username' field from the
authentication line instead of the 'from' field.
Realm
Realm for digest authentication. Realms MUST be globally unique
according to RFC 3261. Set this to your host name or domain name.
Use Domain as
Realm
Use the domain from the SIP Domains setting as the realm. In this case,
the realm will be based on the request 'to' or 'from' header and should
match one of the domain. Otherwise, the configured 'realm' value will be
used.
Always Auth
Reject
When an incoming INVITE or REGISTER is to be rejected, for any reason,
always reject with an identical response equivalent to valid username and
invalid password/hash instead of letting the requester know whether
there was a matching user or peer for their request. This reduces the
ability of an attacker to scan for valid SIP usernames. This option is set to
'yes' by default.
Authenticate
Options Requests
Enabling this option will authenticate OPTIONS requests just like INVITE
requests are. By default this option is disabled.
Allow Guest Calling
Allow or reject guest calls (default is yes, to allow). If your gateway is
connected to the Internet and you allow guest calls, you want to check
which services you offer everyone out there, by enabling them in the
default context.
Media
Table 4-3-6 Instruction of Media
Options
Definition
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Premature Media
Some ISDN links send empty media frames before the call is in ringing or
progress state. The SIP channel will then send 183 indicating early media
which will be empty - thus users get no ring signal. Setting this to "yes" will
stop any media before we have call progress (meaning the SIP channel will
not send 183 Session Progress for early media). Default is 'yes'. Also make
sure that the SIP peer is configured with progressinband=never. In order
for 'noanswer' applications to work, you need to run the progress()
application in the priority before the app.
TOS for SIP
Packets
Sets type of service for SIP packets
TOS for RTP
Packets
Sets type of service for RTP packets
5. Network, Advanced and Logs
Network
On “Network” page, there are “Network Settings”, “DDNS Settings”, and “Toolkit”.
Network Settings
There are three types of LAN port IP, Factory, Static and DHCP. Factory is the default type, and it is
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.
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Figure 5-1-1 LAN Settings Interface
Table 5-1-1 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 5-1-2 DNS Interface
Table 5-1-2 Definition of DNS Settings
Options
Definition
DNS Servers
A list of DNS IP address. Basically this info is from your local
network service provider.
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OpenVPN Settings
You can upload the OpenVPN client configuration, if success, you can see a VPN virtual network
card on SYSTEM status page. About the configure format you can refer to the Notice and Sample
configuration.
Figure 5-1-3 OpenVPN Interface
DDNS Settings
You can enable or disable DDNS (dynamic domain name server).
Figure 5-1-4 DDNS Interface
Table 5-1-3 Definition of DDNS Settings
Options
Definition
DDNS
Enable/Disable DDNS(dynamic domain name server)
Type
Set the type of DDNS server.
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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 5-1-5 Network Connectivity Checking
Advanced
Asterisk API
When you make “Enable” switch to “on”, this page is available.
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Figure 5-2-1 API Interface
Table 5-2-1 Definition of Asterisk API
Options
Definition
Port
Network port number
Manager Name
Name of the manager without space
Manager secret
Password for the manager.
Characters: Allowed characters “-_+.<>&0-9a-zA-Z”. Length:4-32
characters.
Deny
If you want to deny many hosts or networks, use char & as
separator. <br/><br/>Example: 0.0.0.0/0.0.0.0 or
192.168.1.0/255.255.255.0&10.0.0.0/255.0.0.0
Permit
If you want to permit many hosts or network, use char & as
separator.<br/><br/>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, <br/>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.)
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Verbose
Verbose information. Read-only. (Defined but not yet used.)
Command
Permission to run CLI commands. Write-only.
Agent
Information about queues and agents and ability to add queue
members to a queue.
User
Permission to send and receive UserEvent.
Config
Ability to read and write configuration files.
DTMF
Receive DTMF events. Read-only.
Reporting
Ability to get information about the system.
CDR
Output of cdr, manager, if loaded. Read-only.
Dialplan
Receive NewExten and Varset events. Read-only.
Originate
Permission to originate new calls. Write-only.
All
Select all or deselect all.
Once you set like the above figure, the host 172.16.123.123/255.255.0.0 is allowed to access the
gateway API. Please refer to the following figure to access the gateway API by putty.
172.16.123.123 is the gateway’s IP, and 5038 is its API port.
Figure 5-2-2 Putty Access
Asterisk CLI
In this page, you are allowed to run Asterisk commands.
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Figure 5-2-3 Asterisk Command Interface
Table 5-2-2 Definition of Asterisk API
Options
Definition
Command
Type your Asterisk CLI commands here to check or debug your
gateway.
If you type “help” or “?” and execute it, the page will show you the executable commands.
Asterisk File Editor
On this page, you are allowed to edit and create configuration files.
Click the file to edit.
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Figure 5-2-4 Configuration Files List
Click “New Configuration File” to create a new configuration file. After editing or creating, please
reload Asterisk.
Logs
On the “Log Settings” page, you should set the related logs on to scan the responding logs page.
For example, set “System Logs” on like the following, then you can turn to “System” page for
system logs, otherwise, system logs is unavailable. And the same with other log pages.
Figure 5-3-1 System Logs Control
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Figure 5-3-2 System Logs Output
Notice: The same to Asterisk Logs and SIP Logs.
Table 5-3-1 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.<br>
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.
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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.
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