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OpenVox Communication Co., Ltd
ET-200X(L) Series Digital Gateway
User Manual
ET 200X(L) Series Digital Gateway User Manual
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OpenVox Communication Co.,Ltd
Address: 10/F, Building 6-A, Baoneng Science and Technology Industrial Park, Longhua
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Technical Support: support@openvox.cn
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Revision History
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Table of Contents
Revision History
.........................................................................................................................................................
3
Legal Information
..........................................................................................................................................................
4
Confidentiality
.............................................................................................................................................................................
4
Disclaimer
....................................................................................................................................................................................
4
Trademarks
..................................................................................................................................................................................
4
1 Overview
....................................................................................................................................................................
8
1.1 What is ET-200X(L)
................................................................................................................................................................
8
1.2 Sample Application
............................................................................................................................................................
8
1.3 Product Appearance
..........................................................................................................................................................
9
1.4 Main Features
..................................................................................................................................................................
10
1.5 Physical Information
........................................................................................................................................................
11
1.6 Software
............................................................................................................................................................................
11
2 System
...................................................................................................................................................................
12
2.1 Status
.................................................................................................................................................................................
12
2.2 Call Status
.........................................................................................................................................................................
13
2.3 Time
...................................................................................................................................................................................
13
2.4 Login Settings
...................................................................................................................................................................
14
2.5 General
..............................................................................................................................................................................
16
2.5.1 Language Settings
........................................................................................................................................................
16
2.5.2 Scheduled Reboot
.......................................................................................................................................................
16
2.6 Tools
...................................................................................................................................................................................
16
2.6.1 Reboot Tools
................................................................................................................................................................
17
2.6.2 Update Firmware
.........................................................................................................................................................
17
2.6.3 Upload and Backup Configuration
..............................................................................................................................
17
2.6.4 Restore Configuration
.................................................................................................................................................
18
2.7 System Information
.........................................................................................................................................................
18
3 T1/E1
........................................................................................................................................................................
18
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3.1 General
..............................................................................................................................................................................
18
3.2 PRI
.....................................................................................................................................................................................
20
3.3 SS7
....................................................................................................................................................................................
23
3.3.1 Link Set Settings
..........................................................................................................................................................
23
3.3.2 Link Settings
................................................................................................................................................................
24
3.3.3 SS7 Configuration file backup and restore
..................................................................................................................
25
3.4 MFC/R2
.............................................................................................................................................................................
25
3.4.1 MFC/R2 Signaling
........................................................................................................................................................
25
3.4.2 Modify R2 variant
........................................................................................................................................................
26
4 VOIP
........................................................................................................................................................................
31
4.1 VOIP ENDPOINTS
...............................................................................................................................................................
31
4.1.1 SIP Endpoints
...............................................................................................................................................................
31
4.1.2 Main Endpoint Settings
...............................................................................................................................................
31
4.1.3 Advanced Registration Options
...................................................................................................................................
34
4.1.4 Call Settings
.................................................................................................................................................................
34
4.1.5 Advanced Timer Settings
.............................................................................................................................................
34
4.1.6 Advanced Signaling Settings
........................................................................................................................................
35
4.2 IAX2 ENDPOINT
.................................................................................................................................................................
37
4.3 ADVANCED SIP SETTINGS
.................................................................................................................................................
39
4.3.1 Networking
..................................................................................................................................................................
39
4.3.2 Advanced NAT Settings
................................................................................................................................................
40
4.3.3 Advanced RTP Settings
................................................................................................................................................
42
4.3.4 Parsing and Compatibility
...........................................................................................................................................
42
4.3.5 Security
........................................................................................................................................................................
43
4.3.6 Media
..........................................................................................................................................................................
44
4.3.7 Codec Settings
.............................................................................................................................................................
44
4.4 Advanced IAX2 Settings
.................................................................................................................................................
45
4.5 Advanced fax setting
.......................................................................................................................................................
48
5 Routing
..................................................................................................................................................................
49
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5.1 Call Routing Rule
.............................................................................................................................................................
49
5.2 Groups
...............................................................................................................................................................................
52
6 Network
...................................................................................................................................................................
53
6.1 WAN/LAN Settings
..........................................................................................................................................................
53
6.2 DDNS Settings
.................................................................................................................................................................
54
6.3 Toolkit
.................................................................................................................................................................................
55
7 Advanced
..................................................................................................................................................................
55
7.1 Asterisk API
.......................................................................................................................................................................
55
7.2 Asterisk CLI
.......................................................................................................................................................................
57
7.3 Asterisk File Editor
...........................................................................................................................................................
58
7.4 Auto Provisioning
.............................................................................................................................................................
59
7.4.1 Preparation
..................................................................................................................................................................
59
7.4.2 Configuring gateway
....................................................................................................................................................
59
7.4.3 Configuring ACS
...........................................................................................................................................................
61
7.4.4 Provisioning example
..................................................................................................................................................
63
7.5 SNMP
................................................................................................................................................................................
67
7.5.1 Parameters in SNMP setting
........................................................................................................................................
67
7.5.2 Activating SNMP
..........................................................................................................................................................
68
7.5.3 Verify SNMP
.................................................................................................................................................................
68
7.6 TR069
................................................................................................................................................................................
70
7.7 Network Capture
..............................................................................................................................................................
72
8 Logs
........................................................................................................................................................................
73
8.1 Log Settings
......................................................................................................................................................................
73
8.2 System log
........................................................................................................................................................................
76
8.3 Asterisk logs
.....................................................................................................................................................................
76
8.4 Call Statistics
....................................................................................................................................................................
77
8.5 System Notice
..................................................................................................................................................................
77
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1 Overview
1.1 What is ET-200X(L)
ET-200X(L) Digital Gateway is an open source asterisk-based VoIP Gateway solution for operators and call
centers. It is a converged media gateway product. This kind of gateway connects traditional telephone
system to IP networks and integrates VoIP PBX with the PSTN seamlessly. With friendly GUI, users may
easily setup their customized Gateway. Also secondary development can be completed through AMI
(Asterisk Management Interface). ‘X’ means the number of T1/E1 port. ‘L’ stands for none hardware
codec —— V100 module.
It is developed with a wide selection of codecs and signaling protocol, including G.711A, G.711U, G.729,
G.722, G.723 and GSM. It supports PRI/SS7/R2 protocol. ET-200X(L) Digital Gateway has good processing
ability and stability and we provides 1/2/4/8 T1/E1 interface for your choice. The T1/E1 gateway will be
100% compatible with Asterisk, Elastix, trixbox, 3CX, FreeSWITCH SIP server and VOS VoIP operating
platform.
1.2 Sample Application
Figure 1-2-1 Topological Graph
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1.3 Product Appearance
The picture below is appearance of ET-2002.
Figure 1-3-1 Product Appearance
Figure 1-3-2 Front Panel
Table 1-3-1 Description of Front Panel
Interface
Function
Color
Work Status
1 USB
USB interface
2 Eth
Ethernet interface
3 port 1-2
E1 / T1 ports
4 HDMI
High definition multimedia interface
5 ON/OFF
On/off button can be used to turn on/off the system.
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6 PWR
Power status indicator
green
On: power is on
Off:power is off
7 RUN
Register indicator
green
Slow blinking(Green 2s and
Flash 0.1s):Work normally
Fast blinking(Green 0.5s
and Flash 0.5s): Work
abnormally
Fast blinking(Green 0.5s
and Flash 0.5s): Work
abnormally
No blinking: Dahdi Error
8 RST
Reset button is used to restore the devices.
1.4 Main Features
Based on Asterisk
®
Editable Asterisk®configuration file
Codecs support: G.711A, G.711U, G.729, G.723, G.722, GSM
Support PRI/SS7/R2 signling
Support 512 routing rules and flexible routing settings
Stable performance, flexible dialing, friendly GUI
Support ports group management
Support call status information
Support T.38/Pass-through fax
Support Auto Provision, SNMP V1/V2c/V3 and TR069
Echo Cancellation
Connect legacy PBX systems to low-cost VoIP services
Connect legacy PBX systems to remote sites over private VoIP links
Connect IP PBX systems to legacy TDM services
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1.5 Physical Information
Table 1-5-1 Description of Physical Information
Model
ET2001
ET2002
ET2004
ET2001L
ET2002L
E1/T1 port
12412
Codec & EC module
yes
no
dimension
100*162.5mm
weight
210g
216g
226g
202g
207g
temperature（℃）
storage：-40~85
operation
：0~70
Operation humidity
10% ~ 90% non-condensing
Max power
12W
1.6 Software
Default IP: 172.16.100.1(Eth)
Username: admin
Password: admin
Notice: Log in
Figure 1-6-1 LOG IN Interface
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2 System
2.1 Status
On the “System Status” page, you will find all Interface status, channels status, SIP, IAX2, Routing
rules, and Network information.
Figure 2-1-1 System Status
Table 2-1-1 Description of System Status
Options
Definition
Interface Status
Show the status of port, include “OK” and “Down”. “Down” means no trunk line
connected; “OK” means the trunk line of port is available.
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Channels
Status
Show the Channels status of port, include "Idle". "Busy". "Disable" and “S
channel”. "Idle" means it is available;
"Busy" means the channel is busy;
"Disable" means it is unavailable;
“S channel” means signaling channel.
2.2 Call Status
The verbose of the system call status will be present on the “Call Status” page. You can select the
specified T1/E1 port which you are care for.
Figure 2-2-1 Verbose of call status
2.3 Time
Table 2-3-1 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 timezone strings.
NTP Server 1
Time server domain or hostname. For example, [0.cn.pool.ntp.org].
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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 2-3-1 Time Settings
You can set your gateway time Sync from NTP or Sync from Client by pressing different buttons.
2.4 Login Settings
Your gateway doesn't have administration role. All you can do here is to reset what new username
and password to manage your gateway. And it has all privileges to operate your gateway. You can modify
“Web Login Settings” and “SSH Login Settings”. If you have changed these settings, you don’t need to
logout, just rewriting your new user name and password will be OK. Also you can specify the web server
port number. Usually the web login default mode is “http and https”. For safety, you can switch to “only
https” mode.
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Table 2-4-1 Description of Web Login Settings
Options
Definition
User Name
Your gateway does not have administration role.
All you can do here is defining the user name and password to manage
your gateway.
And it has all privileges to operate your gateway .User Name: Allowed
characters “-_+<>&0-9a-zA-Z”.Length:1-32 characters.
Password
Allowed characters "-_+. <>&0-9a-zA-Z".
Length: 4-32 characters.
Confirm Password
Please input the same password as 'Password' above.
Login Mode
Specify the web login mode: http and https, only https. Default is http
and https.
Port
Specify the web server port number. Do not use port 443 which is
reserved for HTTPS.
Figure 2-4-1 Login Settings
Notice: Whenever you do some changes, do not forget to save your configuration.
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2.5 General
2.5.1 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”.
Figure 2-5-1 Language Settings
2.5.2 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 2-5-2 Reboot Types
If use your system frequently, you can set this enable, it can helps system work more efficient.
2.6 Tools
On the “Tools” pages, there are reboot tools, update Firmware, upload Configuration, backup
Configuration and Restore Configuration toolkits.
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2.6.1 Reboot Tools
You can choose system reboot and Asterisk reboot separately.
Figure 2-6-1 Reboot Prompt
If you press “OK”, your system will reboot and all current calls will be dropped. Asterisk Reboot is the
same.
Table 2-6-1 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.
2.6.2 Update Firmware
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, if you choose that, you will see some
information below.
Figure 2-6-2 Prompt Information
2.6.3 Upload and Backup Configuration
If you want to update your system and remain your previous configuration, you can first backup
configuration, then you can upload configuration directly. That will be very convenient for you.
Figure 2-6-3 Upload and Backup
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2.6.4 Restore Configuration
Sometimes there is something wrong with your gateway that you don’t know how to solve it, mostly
you will select factory reset. Then you just need to press a button, your gateway will be reset to the factory
status.
Figure 2-6-4 Factory Reset
2.7 System Information
On the “Information” page, there shows some basic information about the T1/E1 gateway. You can
see software and hardware version, storage usage, memory usage and some help information.
Figure 2-7-1 System Information
3 T1/E1
3.1 General
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Figure 3-1-1 General Settings
Table 3-1-1 Definition of General Settings
Figure 3-1-2 Advanced interface type
Table 3-1-2 Definition of advanced interface type
Figure 3-1-2 Port Details
Table 3-1-3 Definition of Port Details
Options
Definition
Timing Source
Timing Source indicate the ports as to which should be used to recover the
clock.(0 for master mode, upper for client mode, small number have higher
Locale
Your locale. This will be used for the tone style. Used when in-call
indications need to be generated such as ring back, busy, congestion,
and other call-oriented inband tone signals.
Interface Type
It shows you the current type of port. It has two type: E1 and T1
Options
Definition
Echo Cancellation
Whether or not to enable echo cancellation
RX Gain
Gain for the RX (receive -into Asterisk)channel.Default:0.0
TX Gain
Gain for the TX (transmit -out of Asterisk Asterisk)channel.Default:0.0
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priority )
Interface
Choose a line type for the interface.
Framing
Framing method for this interface
Coding
Coding method for this interface
Line Build-out
Line build-out represents the length of the cable form the port on this gateway to
the next device.
CRC4
Enable cyclic redundancy checking for error checking on line. CRC-4 support is
required for all network switches in Europe, but many older switches and PBXs
don’t support it.
Signaling
It shows you what signaling the port uses.
Switch Type
Only used for PRI
Description
An optional description of this interface to be used for reference only.
3.2 PRI
Figure 3-2-2 ISDN: Signaling
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Table 3-2-2 Definition of Signaling
Options
Definition
Q.SIG Channel Mapping
Sets logical or physical channel mapping. In logical channel mapping,
channels are mapped to 1-30.In physical channel mapping, channels are
mapped to 1-15,17-31,skipping the number used for the data channel,
Default is physical.
Enable Caller ID
Whether or not to use caller ID
PRI Dial Plan for Dialed Number
PRI Dialplan: The ISDN-level Type of Number (TON) or numbering plan,
used for the dialed number. Leaving this as ‘unknown’ (the default) works
for most cases. In some very unusual circumstances, you may need to
set this to; ’dynamic’ or ‘redundant’
PRI Dial Plan for Dialing Number
PRI Local Dialplan: Only RARELY used for PRI(sets the calling number’s
numbering plan).In North America, the typical use is sending the 10 digit;
caller ID number and setting the prilocaldialplan to ‘national’ (the
default).Only VERY rarely will you need to change this.
Network Specific Facility (NSF)
Messages
Some switches (AT&T especially) require network specific facility IE.
Supported values are currently ‘none’, ’sdn’, ’megacom’, ’
tollfreemgacom’, ’ account’
Idle Bearer Reset
Whether or not to reset unused B channels
Idle Bearer Reset Period
Sets the time in seconds between restart of unused B channels; defaults
to ‘never’
Display Send
Send/receive ISDN display IE options, the display options are a comma
separated list of the following options:
block: Do not pass display text data.
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name_initial: Use display text in SETUP/CONNECT messages as the
party name.
name_update: Use display text in other messages
(NOTIFY/FACILITY)for COLP name update.
name: Combined name_ initial and name_ update options.
text: Pass any unused display text data as an arbitrary display message
during a call. Sent text goes out in default to ‘name’.
Display Receive
Send/receive ISDN display IE options. The display options are a comma
separated list of the following options:
block: Do not pass display text data.
name_initial: Use display text in SETUP/CONNECT messages as the
party name.
bame_update: Use display text in other messages
(NOTIFY/FACILITY)for COLP name update.
name: Combined name_ initial and name_ update options.
text: Pass any unused display text data as an arbitrary display message
during a call. Sent text goes out in default to ‘name’.
Overlap Dialing
Enable overlap dialing mode--sending overlap digits.
Allow Progress When Call
Released
Allow inband audio (progress) when a call is DISCONNECT Ted by the
end of a PRI
Out-of-Band Indications
PRI Out of band indications. Enable this to report Busy and congestion
on a PRI using out_ of_ band notification. Inband indication, as used by
the gateway doesn’t seem to work with all telcos.
Facility-based ISDN
Supplementary Services
To enable transmission of facility-based ISDN supplementary services
(such as caller name form CPE over facility), enable this option. Cannot
be changed on a reload.
Exclusive Channel Selection
If you need to override the existing channels selection routine and force
all PRI channels to be marked as exclusively selected, set this to yes.
priexclusive cannot be changed on a reload.
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Ignore Remote Hold Indications
If you wish to ignore remote hold indications (and use MOH that is
supplied over the B channel) enable this option.
Block Outbound Caller ID Name
Enable if you need to hide just the name and the number for legacy PBX
use. Only applies to PRI channels.
Wait for Caller ID Name
Support caller ID on call waiting
3.3 SS7
3.3.1 Link Set Settings
Figure 3-3-1 Link Set Settings
You can click button as shown below, when there are several linkset, only one can be set to the
default.
Figure 3-3-2 SS7 Link Set Settings
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Table 3-3-1 Definition of SS7 Link Set Settings
options
Definition
Name
The linkset name
Type
SS7 variant
Called NAI
SS7 Called Nature of Address Indicator
Calling NAI
SS7 Calling Nature of Address Indicator
Network Indicator
What the MTP3 network indicator bits should be set to
International Prefix
International Prefix
National Prefix
National Prefix
Subscriber Prefix
Subscriber Prefix
Unknown Prefix
Unknown Prefix
Point Code
Origin point code
Adj. Point Code
Point code of node adjacent to this signalling link,Possibly the STP between you and
your destination
Default DPC
Default DPC
Sig Chan
Signaling Channel
3.3.2 Link Settings
Figure 3-3-3 Link Settings
You can click button as shown below.
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Figure 3-3-4 SS7 Edit Link Settings
Table 3-3-2 Definition of SS7 Edit Link Settings
options
Definition
Link Index
The link index
Interface Type
T1/E1 mode
Linkset Index
The link member set index
Channel
The voice channel on the link
Port
The T1/E1 port in used
3.3.3 SS7 Configuration file backup and restore
Figure 3-3-5 Configuration file backup and restore
3.4 MFC/R2
3.4.1 MFC/R2 Signaling
Figure 3-4-1 MFC/R2 Signaling
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Table 3-4-1 Definition of MFC/R2 Signaling
options
Definition
Enable Caller ID
Whether or not to use caller ID
Init CAS Bit
The initial position of the CAS bits.
Variant
The standard of MFCR2: ITU, ANSI and China
3.4.2 Modify R2 variant
Figure 3-4-2 R2 Variant
You can click button, then you could fine the below.
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Figure 3-4-3 General
Table 3-4-2 Definition of General
Options
Definition
Variant Name
The variant name
R2 Category
national subscriber works just fine usually
Allow Collect
Default is to block collect calls
Accept On Offer
With this set to ‘no’ then the call will NOT be accepted on offered, and the call will start
irs execution in extensions. Conf until the channel is answered.
Forced Release
Brazil use a special signal to force the release of the line instead of the normal clear
back signal
Charge Calls
Whether or not report to the other end ‘accept call with charge’, when interconnecting
with old PBXs this may be useful
Max DNIS
Max amount of DNIS to ask for
Max ANI
Max amount of ANI to ask for
Get ANI First
Whether or not get the ANI before getting DNIS
Immediate
Accept
This feature allows to skip the use of Group B/II signals and go directly to the accepted
state for incoming calls
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Double Answer
This will cause that every answer signal is changed by answer->clear back->answer,
sort of flash
Skip Category
Skip request of calling party category and ANI
CASNonR2 Bits
Which bits are never used
CAS_R2_Bits
Which bits will be used
Figure 3-4-4 Timer
Table 3-4-3 Definition of Timer
Options
Definition
MF Back Cycle
Max amount of time our backward MF signal can last
MF Back Resume Cycle
Amount of time we set MF signal ON to resume the MF cycle with
MF Fwd Safety
Safety FORWARD timer
R2 Seize
How much time do we wait for a response to our seize signal
R2 Answer
How much to wait for an answer once the call has been accepted
Metering Pulse
Hoe much to wait for metering pulse detection
R2 Double Answer
Interval between ANSWER-CLEAR BACK-ANSWER when double
R2 Answer Delay
Minimum delay time between the Accept tone signal and the R2
CAS Persistence Check
Time to wait for to CAS signaling before handing the new signal
DTMF Start Dial
Safety time before starting to dial DTMF
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DTMF Detection End
Safety time to decide when to stop detecting DTMF DNIS.
Figure 3-4-5 Group A
Figure 3-4-6 Group B
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Figure 3-5-7 Group C
Figure 3-4-8 Group 1
Figure 3-4-9 Group 2
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4 VOIP
4.1 VOIP Endpoints
4.1.1 SIP Endpoints
This page shows everything about your SIP, you can see status of each SIP.
Figure 4-1-1 SIP Status
4.1.2 Main Endpoint Settings
You can click button to add a new SIP endpoint, and if you want to modify existed
endpoints, you can click button.
There are three kinds of registration types for choose. You can choose Anonymous, Endpoint
registers with this gateway or This gateway registers with the endpoint.
You can configure as follows:
If you set up a SIP endpoint by registration “None” to a server, then you can’t register other SIP
endpoints to this server. (If you add other SIP endpoints, this will cause Out-band Routes and Trunks
confused.)
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Figure 4-1-2 None Registration
For convenience, we have designed a method that you can register your SIP endpoint to your
gateway, thus your gateway just work as a server.
Figure 4-1-3 Endpoint Register with Gateway
Also you can choose registration by “This gateway registers with the endpoint”, it’s the same with
“None”, except name and password.
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Figure 4-1-4 This Gateway Register with the Endpoint
Table 4-1-1 Definition of SIP Options
Options
Definition
Name
A name which is able to read by human. And it’s only used for
user’s reference.
Username
User name the end point use to authenticate with the gateway
Password
Password the endpoint will use to authenticate with the gateway.
Allowed characters
Registration
Whether this endpoint will registers with this gateway.
Hostname or IP Address
IP address or hostname of the endpoint or 'dynamic' if the endpoint
has a dynamic IP address. This will require registration. Notice: if
the input here is hostname and your DNS has changed, you must
reboot asterisk.
Transport
This sets the possible transport types for outgoing. Order of usage,
when the respective transport protocols are enabled, is UDP, TCP,
TLS. The first enabled transport type is only used for outbound
messages until a Registration takes place. During the peer
Registration the transport type may change to another supported
type if the peer requests so.
NAT Traversal
Addresses NAT-related issues in incoming SIP or media sessions.
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4.1.3 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.
Qualify
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.
4.1.4 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);
Trust Remote-Party-ID
Whether or not the Remote-Party-ID header should be trusted.
Send Remote-Party-ID
Whether or not to send the Remote-Party-ID header.
Caller ID Presentation
Whether or not to display Caller ID.
4.1.5 Advanced Timer Settings
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Table 4-1-4 Definition of Timer Options
Options
Definition
Default T1 Timer
This timer is used primarily in INVITE transactions. The default for
Timer T1 is 500ms or the measured run-trip time between the gateway
and the device if you have qualify=yes for the 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
Minimum session refresh interval in seconds. Default is 90secs.
Maximum
Session Refresh
Maximum session refresh interval in seconds. Defaults to 1800s.
Session Refresher
The session refresher, uac or uas. Defaults to uas.
4.1.6 Advanced Signaling Settings
Table 4-1-5 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.
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 changes. Turn This option off to
force the SDP session version number and treat all SDP data as new data. This is
require for devices that send non-standard SDP packets (observed with Microsoft OC
S).By default
This option is on.
Allow Transfers
Whether or not to globally enable transfers. Choosing ‘no’ will disable all transfers
(unless enable in peers or users). Default is enabled.
Allow 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 100 Trying when the endpoint registers.
Table 4-1-6 Definition of Fax Options
Options
Definition
Mode
Working mode T.38 and T.30
Enabled
Enabled
Error Correction
Error Correction
Max Datagram
In some cases,T.38 endpoints will provide a T38FaxMxDatagram value (during T.38
setup) that is based on an incorrect interpretation of the T.38 recommendation, and result
in failures because Asterisk does not believe it can send T.38 packets of a reasonable
size to that endpoint (Cisco media gateway are one example of this situation).In these
cases, during a T.38 call you will see warring messages on The console/in the logs from
the Asterisk UDPTL stack complaining about lack of buffer space to send
T.38FaxMaxDatagram value specified by the other end[point, and use a configured value
instead.
Fax Detect
FAX detection will cause the SIP channel to jump to the ‘faX’ extension (if exists) based
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one or more events being detected. The events that can be detected are an incoming
CNG tone or an incoming T.38 re-INVITE request.
Fax Activity
activate T38 fax gateway with ‘timeout’ seconds
Fax Timeout
activate T38 fax gateway with ‘timeout’ seconds
4.2 IAX2 Endpoint
Figure 4-2-1 IAX2 Endpoint
You can click button as shown below
Figure 4-2-2 Edit IAX Endpoint "9001"
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Table 4-2-1 Definition of IAX2 Endpoint
Options
Definition
Name
A name which is able to read by human. And it’s only used for user’s reference.
User name
User name the endpoint will use to authenticate with the gateway
Password
Password the endpoint will use to authenticate with gateway.
Allowed characters
Registration
Whether this endpoint will register to this gateway or this gateway to the endpoint.
Hostname or IP
Address
IP address or hostname of the endpoint or ‘dynamic’ if the endpoint has a dynamic IP
address. This will require registration.
Notice: If the input here is hostname and your DNS has changed, you must reboot
asterisk.
Auth
Authentication method for connections
Transfer
Disable or not IAX2 native transfer
Trunk
Use IAX2 trunking with this host
Qualify
Whether or not to check the endpoint’s connection status.
Qualify Smothing
Use an average of the last two PONG result to reduce falsely detected LAGGED host.
The default is ‘no’.
Qualify Freq Ok
How frequently to ping the peer when everything seems to be OK, in milliseconds.
Qualify Freq not Ok
How frequently to ping the peer when it’s either;
LAGGED or UNAVAILABLE, in milliseconds.
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Port
The port number the gateway will connect to at this endpoint.
Encryption
Enable IAX2 encryption. The default is no.
Force Encryption
Force encryption insures no connection is established unless both sides support
encryption. By turning this option on, encryption is automatically; turned on as well. The
default is no.
Trunk Max Size
Defaults to 128000 bytes, which supports up to 800; calls of ulaw at 20ms a frame.
Trunk MTU
With a large amount of traffic on IAX2 trunk, there is a risk of bad voice quality when
allowing the Linux system to handle fragmentation of UDP packets.
Depending on the side of each payload, allowing the OS to handle fragmentation may
not be very efficient. This setting sets the maximum transmission unit for AIX2 UDP
trunking. The default is 1240 bytes which means if a trunk’s payload is over 1240 bytes
for every 20ms it will be broken into multiple 1240 bytes messages. Zero disables this
functionality and let’s the OS handle fragmentation.
Trunk Frequency
How frequently to send trunk msgs (in ms). This is 20ms by default.
Trunk Time Stamps
Should we send timestamps for the individual sub_frames within trunk frames? There is
a small bandwith use for these (less than 1kbps/call), but they ensure that frame
timestamps get sent end-to-end properly. If both ends of all your trunks go directly to
TDM, _and_your trunkfreq equals the frame length for your codecs, you can probably
suppress these. The receiver must also need to have it enabled.
Min. RegExpire
Minimum amounts of time that IAX2 peers can request as a registration interval (in
seconds).
Max. RegExpire
Maximum amounts of time that IAX2 peers can request as a registration expiration
interval (in seconds).
4.3 Advanced SIP Settings
4.3.1 Networking
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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.
4.3.2 Advanced NAT Settings
Table 4-3-2 Definition of NAT Settings Options
Options
Definition
Local Network
Format:192.168.0.0/255.255.0.0 or 172.16.0.0./12. A list of IP address or IP
ranges which are located inside a NATed network. This gateway will replace
the internal IP address in SIP and SDP messages with the external IP address
when a NAT exists between the gateway and other endpoints.
Local Network List
Local IP address list that you added.
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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
Only substitute the exeternaddr or externhost setting if it matches
Dynamic Exclude
Static
Disallow all dynamic hosts from registering as any IP address used for staticly
defined hosts .This helps avoid the configuration error of allowing your users to
register at the same address as a SIP provide.
Externally Mapped
TCP Port
The externally mapped TCP port, when the gateway is behind a static NAT or
PAI
External Address
The external address (and optional TCP port) of the NAT. External
address=hostname [:port] specifies a static address[:port] to be used in SIP
and SDP messages. Examples: External address=12.34.56.78 External
address=12.34.56.78.9900
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.
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4.3.3 Advanced RTP Settings
Table 4-3-3 Definition of RTP Settings Options
Options
Definition
Start of RTP Port Range
Start of range of port numbers to be used for RTP.
End of RTP port Range
End of range of port numbers to be used for RTP.
4.3.4 Parsing and Compatibility
Table 4-3-4 Instruction of Parsing and Compatibility
Options
Definition
Strict RFC
Interpretation
Check header tags, character conversion in URIs, and multiline
headers for strict SIP compatibility(default is yes)
Send Compact
Headers
Send compact SIP headers
SDP Owner
Allows you to change the username filed in the SDP owner string.
This filed MUST NOT contain spaces.
Disallowed SIP
Methods
When a dialog is started with another SIP endpoint, the other endpoint
should include an Allow header telling us what SIP methods the
endpoint implements. However, some endpoint either do not include an
Allow header or lie about what methods they implement. In the former
case, the gateway makes the assumption that the endpoint support all
known SIP methods. If you know that your SIP endpoint does not
provide support for a specific method, then you may provide a list of
methods that your endpoint does not implement in the disallowed_
methods option. Note that if your endpoint is truthful with its Allow
header, then there is need to set this option.
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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
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.
4.3.5 Security
Table 4-3-5 Instruction of Security
Option
Definition
Match Auth
Username
If available, match user entry using the 'username' field from the
authentication line instead of the 'from' field.
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.
4.3.6 Media
Table 4-3-6 Instruction of Media
Options
Definition
TOS for SIP Packets
Sets type of service for SIP packets
TOS for RTP Packets
Sets type of service for RTP packets
4.3.7 Codec Settings
Select codecs from the list below.
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Figure 4-3-1 Codec Settings
4.4 Advanced IAX2 Settings
Table 4-4-1 Instruction of General
Options
Definition
Bind Port
Bind port and bindaddr may be specified
Enable IAXCompat
More than once to bind to multiple addresses, but the first will be the default.
Enable
Nochecksums
Set iaxcompat to yes if you plan to use layered switches or some other scenario which
may cause some delay when doing a lookup in the dialplan. It incurs a small
performance hit to enable it. This option cause Asterisk to spawn a separate thread
when it receives an IAX DPREQ (Dialplan Request) instead of blocking while it waits
for a response.
Enable Delay Reject
Disable UDP checksums (if no checksums is set, then no checksums will be
calculated/checked on system supporting the feature)
ADSI
ADSI (Analog Display Services Interface) can be enable if you have (or may have)
ADSI compatible CPE equipment.
SRV Loopup
Whether or not to perform an SRV lookup on outbound calls
AMA Flags
You may specify a global default AMA flag for iaxtel calls. These flags are used in the
generation of call detail records.
autokill
If we don’t get ACK to our NEW within 2000ms,and autokill is set to yes, then we
cancel the whole thing(that’s enough time for one retransmission only ).This is used to
keep things from stalling for a long time for a host that is not available for bad
connections.
Language
You may specify a global default language for users. This can be specified also on a
per-user basis. If omitted, will fallback to English(en)
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Account Code
You may specify a default account for Call Detail Records (CDRs) in addition
specifying on a per-user basis.
Table 4-4-2 Instruction of Music on Hold
Options
Definition
Mohsuggest
The ‘Mohsuggest’ option specifies which music on hold class to suggest to the peer
channel when this channel place the peer on hold. It may be specified globally or on a
per-user or per-peer basis.
Mohinterpret
You may specify a global default language for users. This can be specified also on a
per-user basis. If omitted, will fall back to English(en)
Table 4-4-3 Instruction of Codec Settings
Options
Definition
Band Width
Specify bandwith of low, medium, or high to control which codes are used in general
Disallow
Fine tune codes here using “allow” and “disallow” clause with specific codes
Allow
Fine tune codes here using “allow” and “disallow” clause with specific codes
Codec Priority
Codec priority controls the codec negotiation of an inbound IAX2 call. This option is
inherited to all user entity separately which will override the setting in general.
Table 4-4-4 Instruction of Jitter Buffer
Options
Definition
Jitter Buffer
Global default as to whether you want the jitter buffer at all
Force Jitter Buffer
In the ideal world, when we bridge VoIP channels we don’t want to jitter buffering on
the switch, since the endpoints can each handle this. However, some endpoints may
have poor jitter buffers themselves, so this option will force to always jitter buffer, even
in this case.
Max Jitter Buffers
A maximum size for the jitter buffer
Resyncthreshold
When the jitter buffer notice a significant change in delay that continue over a few
frames, it will resync, assuming that the change in delay was caused by a
timestamping mix-up. The threshold for noticing a change in delay is measured as
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twice the measured jitter plus this resync threshold.
Max Jitter Interps
The maximum number of interpolation frames the jitter buffer should return in a row.
Since some clients do not send CNG/DTX frames to indicate silence, the jitter buffer
will assume silence has begun after returning this many interpolations. This prevents
interpolating throughout a long silence.
Jitter Target Extra
Number of milliseconds by which the new jitter buffer will pad its size. The default is
40, so without modification, the new jitter buffer will set its size to the jitter value may
help if your network normally has low jitter, but occasionally has spikes.
Table 4-4-5 Instruction of Misc Settings
Options
Definition
IAX Thread Count
Establishes the number of iax helper thread to handle I/O
IAX Max Thread Count
Establishes the number of extra dynamic threads that may by spawned to
handle I/O
Max Call Number
The ‘maxcallnumbers’ option limits the amount of call numbers allowed for
each individual remote IP address. Once an IP address reaches its call
number limit, no more new connections are allowed until the previous ones
close. This option can be used in a peer definition as well, but only takes
effect for the IP of a dynamic peer after it completes registration.
MaxCallNumbers_Nonvalidated
The ‘maxcallnumbers-nonvalidated’ is used to set the combined number of
call numbers that can be allocated for connections where call token
validation has been disabled. Unlike the ‘maxcallnumbers’ option, this limit
is not separate for each individual IP address. Any connection resulting in a
non-call token validated call number being allocated contributes to this
limit. For use cases, see the call should be sufficient in most cases.
Table 4-4-6 Instruction of Quality of Service
Options
Definition
Tos
Type of service
Cos
Class of service
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4.5 Advanced fax setting
Table 4-5-1 Instruction of Quality of Fax Settings
Options
Definition
Udptl Start
DPTL start configure addresses
Udptl End
DPTL end configure addresses
Udptl Checksums
Whether to enable or disable UDP checksums on UDPTL traffic
Udptl Fec Entries
The number of error correction entries in a UDPTL packet
Udptl Fec Span
The span over which parity is calculated for FEC in a UDPTL packet
Use Even Ports
Some VoIP providers will only accept an offer with an even-numbered UDPTL port.
Set this option so that Asterisk will only attempt to use even-numbered ports when
negotiating T.38. Default is no.
Maximum Transmission
Rate
Maximum Transmission Rate
Minimum Transmission
Rate
Minimum Transmission Rate
Send Progress/Status
events to manager
session
Manager events with ‘call’ class permissions will receive events indicating the
steps to initiate a fax session. Fax completion events are always sent to manager
sessions with ‘call’ class permissions, regardless of the value of this option.
Modem Capabilities
Set this value to modify the default modem options. Defasult:v17,v27,v29
ECM
Enable/disable T.30 ECM(error correction mode) by default
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5 Routing
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 usage of DID function: How to use DID function with OpenVox T1/E1 Gateway). The gateway
support trunk group and trunk priority management.
5.1 Call Routing Rule
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
b
utton 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 5-1-2
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
288N/A
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You can click button to set up your routings.
Figure 5-1-3 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 5-1-1 Definition of Routing Options
Options
Definition
Routing Name
The name of this route. Should be used to describe what types of calls
this route matches (for example, 'SIP2Ports' or 'Ports2SIP').
Call Comes in From
The launching point of incoming calls.
Send call Through
The destination to receive the incoming calls.
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Table 5-1-2 Description of Advanced Routing Rule
Options
Definition
Dial Patterns that will
use this Route
A Dial Pattern is a unique set of digits that will select this route and send
the call to the designated trunks. If a dialed pattern matches this route,
no subsequent routes will be tried. If Time Groups are enabled,
subsequent routes will be checked for matches outside of the
designated time(s).
Rules:
X matches any digit from 0-9
Z matches any digit from 1-9
N matches any digit from 2-9
[1237-9] matche any digit in the brackets (example: 1,2,3,7,8,9)
wildcard* : matches one or more dialed digits.
prepend: Digits to prepend to a successful match. If the dialed number
matches the patterns specified by the subsequent columns, then this will
be prepended before sending to the trunks.
prefix: Prefix to remove on a successful match. The dialed number is
compared to this and the subsequent columns for a match. Upon a
match, this prefix is removed from the dialed number before sending it to
the trunks.
match pattern: The dialed number will be compared against the prefix
+ this match pattern. Upon a match, the match pattern portion of the
dialed number will be sent to the trunks
SDfR(Stripped Digits from Right): The amount of digits to be deleted
from the right end of the number. If the value of this item exceeds the
length of the current number, the whole number will be deleted.
RDfR( Reserved Digits from Right) :Designated information to be added
to the right end of the current number.
StA(Suffix to Add):Designated information to be added to the right end
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of the current number.
Caller Name: What caller name would you like to set before sending
this call to the endpoint. Native language charset is allowable, e.g.
Chinese charset, Latin charset.
Forward Number
What destination number will you dial?
This is very useful when you have a transfer call.
Failover Call Through
Number
The gateway will attempt to send the call out each of these in the order
you specify.
You can create various time routes and use these time conditions to limit some specific calls.
Figure 5-1-4 Time Patterns that will use this Route
If you configure like this, then from January to March, from the first day to the last day of these
months, from Monday to Thursday, from 00:00 to 02:00, during this time (meet all above time conditions),
all calls will follow this route. And the time will synchronize with your Sever time.
Figure 5-1-5 Forward number
You can also configure forward number when you have a transfer call.
Figure 5-1-6 Failover Call Through Number
You can add one or more “Failover Call Through Numbers”.
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
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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 Establish Group
6 Network
On “Network” page, there are three sub-pages, “WAN Settings”, “DDNS Settings” and “Toolkit”.
6.1 WAN/LAN Settings
There are two types of WAN port IP, Static and DHCP. Static is the default type, and it is 172.16.100.1.
The LAN port is a fixed IP and it is 192.168.100.1.
Figure 6-1-1 WAN/LAN Settings Interface
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Table 6-1-1 Definition of WAN/LAN Settings
Options
Definition
Interface
The name of network interface.
Type
The method to get IP.
Static: manually set up your gateway IP.
MAC
Physical address of your network interface.
Address
The IP address of your gateway.
Network
The subnet mask of your gateway.
Default Gateway
Default getaway IP address.
DNS Servers: A list of DNS IP address. Basically this info is from your local network service provider.
Note that please restart the gateway if you changed the DNS server.
Figure 6-1-2 DNS Interface
6.2 DDNS Settings
You can enable or disable DDNS (dynamic domain name server).
Figure 6-2-1 DDNS Interface
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Table 6-2-1 Definition of DDNS Settings
Options
Definition
DDNS
Enable/Disable DDNS(dynamic domain name server)
Type
Set the type of DDNS server.
Username
Your DDNS account’s login name.
Password
Your DDNS account’s password.
Your domain
The domain to which your web server will belong.
6.3 Toolkit
It is used to check network connectivity. Support Ping command on web GUI.
Figure 6-3-1 Network Connectivity Checking
7 Advanced
7.1 Asterisk API
When you make “Enable” switch to “ON”, this page is available.
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Figure 7-1-1 API Interface
Table 7-1-1 Definition of Asterisk API
Options
Definition
Port
Network port number
Manager Name
Name of the manager without space
Manager secret
Password for the manager.
Characters: Allowed characters “-_+.<>&0-9a-zA-Z”. Length:4-32 characters.
Deny
If you want to deny many hosts or networks, use char & as separator. Example:
0.0.0.0/0.0.0.0 or 192.168.1.0/255.255.255.0&10.0.0.0/255.0.0.0
Permit
If you want to permit many hosts or network, use char & as separator. Example:
0.0.0.0/0.0.0.0 or 192.168.1.0/255.255.255.0&10.0.0.0/255.0.0.0
System
General information about the system and ability to run system management
commands, such as Shutdown, Restart, and Reload.
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.
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Config
Ability to read and write configuration files.
DTMF
Receive DTMF events. Read-only.
Reporting
Ability to get information about the system.
Dialplan
Receive NewExten and Var Set events. Read-only.
Originate
Permission to originate new calls. Write-only.
All
Select all or deselect all.
Once you set like the above figure, the host 172.16.100.110/255.255.0.0 is allowed to access the
gateway API. Please refer to the following figure to access the gateway API by putty. 172.16.100.110 is
the gateway’s IP, and 5038 is its API port.
Figure 7-1-2 Putty Access
7.2 Asterisk CLI
In this page, you are allowed to run Asterisk commands.
Figure 7-2-1 Asterisk Command Interface
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Table 7-2-1 Definition of Asterisk CLI
Options
Definition
Command
Type your Asterisk CLI commands here to check or debug your
If you type “help” or “?” and execute it, the page will show you the executable commands.
Table 7-2-2 Definition of Lock/unlock channels
Options
Definition
Signaling
Current signaling in use
Operation
The advanced operations for lock and unlock channels
Channel:
The channel to be lock or unlock
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
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reload Asterisk.
7.4 Auto Provisioning
Auto provisioning or auto-configuration is an easy, flexible and time-saving way to upgrade firmware
and configurations for E1 gateways in mass deployment. With auto provisioning, all user information can
be entered via the central ACS (Auto Configuration Server). ACS can be DHCP server or TFTP, HTTP
and FTP server. It will not take effects immediately but in the next time system is power on. It could be
postponed the execution of restart system also.
Note that system will not be upgrade the firmware and update configurations if the connection
between ACS and gateway is disconnect.
7.4.1 Preparation
The following should be prepared before anto provisioning being applied.
Enable the auto provisioning in gateway
The ACS has been prepared
The network between gateway and ACS is connected
7.4.2 Configuring gateway
Usually, the feature is disabled before being on sale. To activate the auto provisioning function,
please follow the procedures as below.
Step 1 On the ADVANCED-> Auto Provision interface
Step 2 Enable the ‘Enabled’ option and select ACS. DHCP option 66 can be enabled if ACS has been
work as DHCP server, otherwise please select protocol of provisioning and fill the value of ‘Auto
Config Server URL’. Username and password may need to be filled in FTP/HTTP for the purpose
of system safety. Do not forget to select Firmware upgrade, upgrade mode and fill the value of
timeout, and click ‘Save’.
Step 3 Set interval of checking in LOGS->System notice then enable it, and click ‘Save’.
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Table 7-4-1 Definition of Auto Provision
Options
Definition
Enabled
Whether to enable or disable Auto Provision
DHCP Option 66
Get ACS server address from Option 66 via DHCP
Protocol
Set protocol of connection
Auto Config Server URL
The config server domain or IP address
User Name
The account of downloading from ACS
Password
The password of downloading from ACS
Timeout
The max limit time for downloading firmware
Firmware Upgrade
Enable/disable the mode of downloading firmware
Upgrade Mode
Select upgrade time.
Power: start upgrade configuration when Power on. Power + Period: Set the
frequency of checking the latest configuration when gateway running
Table 7-4-2 Definition of system notice
Options
Definition
Enable
Whether to enable or disable system notice
Check Interval
When Upgrade Mode is set, this parameter specifies the interval of Checking.
Figure 7-4-1 Auto Provision interface
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7.4.3 Configuring ACS
The Auto Configuration Server can be the one of TFTP, FTP and HTTP server. The ACS is used to
store the firmware release and configurations files of the devices under management.
List the primary files in ACS download directory as table 7-4-3:
Table 7-4-3 Definition of ACS files
Options
Definition
DGW100x-current.bin
The firmware image
common.conf
The wildcard configuration file for the whole gateway
defconfig.tar.gz
The default(factory) configuration file
EPC-{mac}.conf
The private configuration file for the specified gateway.
Naming rules: “EPC-“ + “mac” +”.conf”. The naming prefix of “EPC-” stands for the
private configuration file, “mac” is the physical address of network interface card but
removed semicolon and “.conf” is the suffix. For example, the EPC-a0980501dbca.conf,
‘a0980501dbca’ is the MAC address (A0:98:05:01:DB:CA).
The format of common.conf , EPC-{mac}.conf and defconfig.tar.gz:
(1). Common.conf
[firmware]
FW_NAME=DGW100x-current.bin //Firmware image name
FW_MD5=b3603f3c3b5e7eb6326498640f151c79 //The md5 of firmware image
FW_VERSION=1.1.2 //Firmware version
[configs]
CONFIG_NAME=defconfig.tar.gz // default configuration file(compressed)
CONFIG_MD5KEY=2cd2dfbe52482405350816e3698cb530 // the md5 of default configuration file
(2).EPC-{mac}.conf
[dns]
DNS_SERVER1=8.8.8.8
DNS_SERVER2=8.8.4.4
DNS_SERVER3=
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DNS_SERVER4=
[ntp]
NTP_SERVER1= 0.cn.pool.ntp.org
NTP_SERVER2= time.nist.gov
NTP_SERVER3= time.windows.com
[eth0]
ENABLE=yes
TYPE=static
DHCP=no
IPADDRESS=172.16.100.223
NETMASK=255.255.0.0
GATEWAY=172.16.0.1
[eth1]
ENABLE=yes
TYPE=static
DHCP=no
IPADDRESS=192.168.100.223
NETMASK=255.255.0.0
GATEWAY=192.168.0.1
[web_login]
username=admin
password=admin
(3). Defconfig.tar.gz
Figure 7-4-2 the overview of defconfig.tar.gz
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7.4.4 Provisioning example
After auto provisioning is enabled, the gateway will visit the Auto Configuration Server and download
the updated files periodically based on the timer Check Interval (LOGS->System notice). By default, the
timer is set as every hour. System will receive a message from ACS, like figure 7-4-3, and the message
will be display in the system notice (LOGS->System Notice).
Auto provisioning will not take effects immediately but in the next time system is power on. It could be
postponed the execution of restart system also.
Figure 7-4-3 Auto provision notice
Now, an example of using Auto Provisioning will be given in the following.
1. Activate the auto provision (TFTP) in ADVANCED-> Auto Provision like figure 7-4-4.
Figure 7-4-4 Auto provision settings
2. Enable the check interval in LOGS->Log settings->System Notice like figure 7-4-5.
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Figure 7-4-5 Check interval setting
3. Configuring the ACS(Generate the md5 of firmware and defconfig.tar.gz)
Copy the firmware, defconfig.tar.gz, common.conf and EPC-{mac}.conf to the working
directory of TFTP server.
Figure 7-4-6 The working directory of TFTP server
Notice:
The demo of E1 gateway mac address is A0:98:05:01:DB:CA (eth0), therefore the private
configuration file is EPC-a0980501dbca.conf.
Generate the md5 of firmware and defconfig.tar.gz. Then fill in common.conf and
EPC-{mac}.config.
Figure 7-4-7 Generate the md5 of firmware and configuration
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Figure 7-4-8 Common.conf
Figure 7-4-9 EPC- a0980501dbca.conf
Start TFTP service. Tftpd32.exe is a useful TFTP tools in windows7, then make sure TFTP
server is select.
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Figure 7-4-10 Demo TFTP server
4. The system will receive an auto provision message in web GUI.
Figure 7-4-11 System notice logs
Figure 7-4-12 Auto provision upgrade notification
5. Restart the system. It will take about 3 minutes almost to download, upgrade Firmware and
update configurations.
Figure 7-4-13 Downloading the firmware and configs
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Figure 7-4-14 Applying the firmware and configs
7.5 SNMP
Simple Network Management Protocol (SNMP) is an application–layer protocol, which is used to
manage and monitor network elements and exchange management information between network devices.
By default SNMP uses port 161 for communication.
Since the inception SNMP, it embraces three versions: v1, v2c and v3. V1 and v2c are the most
implemented version of SNMP; v3 is target at the high security when compare to its older versions. The
gateway support private SNMP MIBs (private enterprise number) to access.
7.5.1 Parameters in SNMP setting
Table 7-5-1 Definition of SNMP setting
Options
Definition
SNMP Enable
Whether to enable SNMP
System Contact
System contact information(optional)
System Location
The locale of system contact(optional)
Private Enterprise Number
The number is used for defining private SNMP MIBs which is assigned by Internet
Assigned Numbers Authority (IANA). For more information, please access:
http://pen.iana.org/pen/PenApplication.page
SNMP Version
Select version of SNMP
Community Configuration
Define a community name to security name
Group Configuration
Define the security name to a group
View Configuration
Set a view to let the group have rights to do
Access Configuration
Grant the group can access to the view(read/write/notify)
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User Configuration
Only exist in v3. Add a v3 account to SNMP. Notice that the length of auth
password and privacy password are more than 8.
7.5.2 Activating SNMP
Usually, the feature is disabled by default. To activate the SNMP feature, please follow the Figure
7-5-1.
The Interface is in the ADVANCED->SNMP. System contact, location and private enterprise number
are optional. Figure 7-5-1 is the SNMP setting interface.
Figure 7-5-1 Activating the SNMP
Note: Do not forget to click ‘Save’ to take effect. After configuration, The SNMP feature is activated
immediately.
7.5.3 Verify SNMP
A powerful, indispensable and easy-to-use MIB browser is convenient for engineer/manager to
manage SNMP enabled network devices and applications. In this session, Manage Engine MIB browser
is selected. It allows user to issue SNMP requests to retrieve agent's data, or make changes to the agent.
It is free tool for Windows, Mac and Linux.
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(1). Get SNMP parameters via SNMP MIB browser. It’s supposed that Manage Engine MIB browser
is installed perfectly. Figure 7-5-2 is the main interface of Manage Engine MIB browser.
Figure 7-5-2 Manage Engine MIB browser
And the field of Host, Port and Community are filled with 172.16.100.223, 161 and public
respectively. Object ID is the node of SNMP MIBs, e.g. “.1.3.6.1.2.1.1.6.0” is system location and
“.1.3.6.1.2.1.1.1.0” is system description.
Figure 7-5-3 Get system location
After the rest field has been filled, then verify it. Click Operations->GET to get the value of system
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location and it returns the value which we just set.
(2). Set SNMP parameters via SNMP MIB browser. For example, set the system name. system name
is “dgw100x” by default, then set it as “VoIP gateway”. See figure 7-5-4.
Click Operations->GET to attain the current system name.
Fill the field of Set Value with “VoIP gateway”.
Click Operations->SET to set the system name.
Click Operations->GET to attain the modified system name.
Figure 7-5-4 Set system name
7.6 TR069
TR069 is a remote management solution which offers a single interface to manage the ACS and
automate the deployment and support of data, voice and video services, thereby reducing operation and
support costs, while enhancing customer satisfaction. Its user-friendly interface covers the entire service
lifecycle, from centralized remote provisioning of services, to inventory management, group updates,
monitoring, event triggering, and support automation. Figure 7-6-1 is TR-069 configuration interface and
table 7-6-1 is its definition.
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Table 7-6-1 Definition of TR069 configuration interface
Options
Definition
Acs Url
Specify the URL of the ACS
Acs Username
Specify the user name to be used by the device to authenticate with the ACS.
Acs Password
Specify the password to be used by the device to authenticate with the file server
Provisioning Code
Information of the device vendor, which may be used to indicate the primary service
provider and other provisioning information to the ACS. It can be numbers or English
letters.
Model Name
A brief description of the interface type or name. It is a string of characters.
Periodic Enable
Used to specify whether to periodically report to the ACS.
Periodic Interval
The interval for reporting to the ACS.
Connection Request Url
The address used for the ACS to connect back to the device.
Connection Request
Username
The account used for the ACS to connect back to the device, for example, admin.
Connection Request
Password
The password used for the ACS to connect back to the device.
Figure 7-6-1 TR069 configuration interface
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7.7 Network Capture
The gateway have been supplied a network packets capture in the web for ease of user to analysis,
capture and monitor the gateway’s network status, RTP flows, protocol analysis and so on.
Table 7-7-1 Definition of Network capture
Options
Definition
Network Interface
Specify which interface to be capture packets from. ‘All’ means
capture packets from all interfaces
Source host
Specify which source host IP address to listen for
Destination host
Specify which destination host IP address to listen for
Port
To specify a port that is either source or destination direction
Protocol
To specify which protocol to be captured, ‘All’ stands for capture
multi-protocols, the SIP default port is 5060, If you are using a
different port, please amend it.
The interface is in ADVANCED->Network Capture.
Figure 7-7-1 Network capture interface
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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 “SIP Logs” on like the following, then you can turn to “SIP” page for sip logs, otherwise,
sip logs is unavailable. And the same with other log pages.
Figure 8-1-1 Logs Settings
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Figure 8-1-2 System Logs Output
Table 8-1-1 Definition of Logs
Options
Definition
Auto clean
(System Logs)
Switch on: when the size of log file reaches the max size,
The system will cut a half of the file. New logs will be retained.
Switch off: logs will remain, and the file size will increase gradually.
Verbose
Asterisk console verbose message switch.
Notice
Asterisk console notice message switch.
Warning
Asterisk console warning message switch.
Debug
Asterisk console debug message switch.
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Error
Asterisk console error message switch.
DTMF
Asterisk console DTMF info switch.
Auto clean:
(asterisk logs)
Switch on: when the size of log file reaches the max size,
The system will cut a half of the file. New logs will be retained.
Switch off: logs will remain, and the file size will increase gradually.
default on, default size=2 MB
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=2 MB
IAX2 Logs
Whether enable or disable IAX log
Auto clean
Switch on: when the size of log file reaches the max size,
The system will cut a half of the file. New logs will be retained.
Switch off: logs will remain, and the file size will increase gradually.
default on, default size=2 MB
MFC/ R2 Logs
Whether enable or disable MFC/ R2 Logs log.
Auto clean
Switch on: when the size of log file reaches the max size,
The system will cut a half of the file. New logs will be retained.
Switch off: logs will remain, and the file size will increase gradually.
default on, default size=2 MB
PRI Logs
PRI port logs. You can choose one or more ports. If you choose "All", the "PRI"
page will show you the logs about all the ports.
Auto clean (PRI logs)
Switch on: when the size of log file reaches the max size,
The system will cut a half of the file. New logs will be retained.
Switch off: logs will remain, and the file size will increase gradually.
default on, default size=2 MB
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.SS7 Logs
Whether enable or disable SS7 log
Auto clean
switch on : when the size of log file reaches the max size,
The system will cut a half of the file. New logs will be retained.
Switch off: logs will remain, and the file size will increase gradually.
default on, default size=2 MB
Call Statistics
Whether enable or disable Call Statistics.
System Notice
The notification from system firmware upgrade and Auto provisioning
8.2 System log
System log record every time power on, power off and firmware upgrade information.
Figure 8-2-1 System Log
8.3 Asterisk logs
On the pages of “Asterisk”, “SIP”, “IAX2”, “SS7”, “PRI” and “MFC/R2”, there owns the some
functions—Displays the log by port, refresh regularly and log download.
Figure 8-3-1 Asterisk Log
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8.4 Call Statistics
The figure of call statistics, you’ll find “Answered”, “congestion”, “Call busy”, “Call failed”, “No
answer”, “Current calls”, “accumulated calls”, “Calls duration” and “ASR”. “ASR” stands for Answer
Seizure Ratio. “Calls duration” will count the whole calls in the gateway. The call statistics will be saved
before power off. It will be loaded after power on. It can be refreshed by itself. You can reset the statistics
manually.
Figure 8-4-1 Call Statistics
Note: Do not forget to enable call statistics in “Log Setting” if you want to statistics the calls.
8.5 System Notice
The system notice could be generated by system to inform the network manager of what is going on if
it has been enabled. Firmware upgrade messages from official website and auto provisioning messages
from ACS are main notice right now. And at first, enable the system notice function like figure 8-5-1.
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Figure 8-5-1 enable system notice function
After about an hour, a system message is received in the web like 8-5-2.
Figure 8-5-2 enable system notice function
Note: Do not forget to enable system notice and check interval in “Log Setting” if you want to
receive system messages.
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