DGW-100XR User Manual

Table of Contents

1.  Overview

What is DGW-100XR?

OpenVox T1/E1 Gateway is an open source asterisk-based VoIP Gateway solution for operators and call centers. It is a converged media gateway product. This kind of gateway connects traditional telephone system to IP networks and integrates VoIP PBX with the PSTN seamlessly. With friendly GUI, users may easily setup their customized Gateway. Also secondary development can be completed through AMI (Asterisk Management Interface). The DGW-100XR could support two power supply and DGW-100X series gateway support one power supply.

It is developed with a wide selection of codecs and signaling protocol, including G.711A, G.711U, G.729, G.722, G.723 and GSM. It supports PRI/SS7/R2 protocol. OpenVox T1/E1 Gateway has good processing ability and stability and we provides 1/2/4 T1/E1 interface for your choice. The T1/E1 gateway will be 100% compatible with Asterisk, Elastix, trixbox, 3CX, FreeSWITCH SIP server and VOS VoIP operating platform.

Sample Application

                                  Figure 1-2-1Topological Graph

Product Appearance

The picture below is appearance of DGW-1004.

                                            Figure 1-3-1 Product Appearance


                                             Figure 1-3-2 Front Panel

             Table 1-3-1 Description of Front Panel




Work Status

1 Port 1-Port4

E1/T1 ports. The port numbers are different on different models, from 1 to 4.

2 Reset

Reset button is used to restore the device.


Register indicator


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


Power Status indicator


On: Power is on

Off: Power is off


VGA monitor connector

6 Eth1

Network interface

7 Eth0

Network interface


USB interface

                                                Figure 1-3-3 Backup Panel

The OpenVox DGW-100X series gateways provides one or two power supply, one power named DGW-100X ,the other is named DGW-100XR, ‘R’ stands for Reduntant.

Main Features

    • Based on Asterisk®
    • Editable Asterisk® configuration file
    • Wide selection of codecs and signaling protocol
    • Support 512 routing rules and flexible routing settings
    • Stable performance, flexible dialing, friendly GUI
    • Codecs support: G.711A, G.711U, G.729, G.723, G.722, GSM
    • Support ports group management
    • 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

Physical Information

   Table 1-5-1 Description of Physical Information


2842 g




-40~85°C (Storage)

0~70°C (Operation)

Operation humidity

5%~95% non-condensing

Max power

20 W

LAN port


WAN port



Default IP:

Username: admin

Password: admin

Notice: Log in

                Figure 1-6-1 LOG IN Interface

2. System


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



Interface Status

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

Channels Status

Show the Channels status of port, include "Idle". "Busy". "Disable" and “S channel”.

"Idle" means it is available;

"Busy" means the channel is busy;

"Disable" means it is unavailable;

“S channel” means signaling channel.


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


             Table 2-2-1Description of Time Settings



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 timezone strings.

NTP Server 1

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

NTP Server 2

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

NTP Server 3

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

Auto-Sync from NTP

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

Sync from NTP

Sync time from NTP server.

Sync from Client

Sync time from local machine.

For example, you can configure like this:

                                             Figure 2-2-1 Time Settings

You can set your gateway time Sync from NTP or Sync from Client by pressing different buttons.

Login Settings

Your gateway doesn't have administration role. All you can do here is to reset what new username and password to manage your gateway. And it has all privileges to operate your gateway. You can modify “Web Login Settings” and “SSH Login Settings”. If you have changed these settings, you don’t need to log out, just rewriting your new user name and password will be OK. Also you can specify the web server port number.Usually the web login default mode is “http and https”. For safety, you can switch to “only https” mode.

                 Table 2-3-1Description of Login Settings



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.


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.


Specify the web server port number. Do not use port 443 which is reserved for HTTPS.

                                                         Figure 2-3-1 Login Settings

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


Language Settings

You can choose different languages for your system. If you want to change language, you can switch “Advanced” on, then “Download” your current language package. After that, you can modify the package with the language you need. Then upload your modified packages, “Choose File” and “Add”.


                                             Figure 2-4-1 Language Settings

Scheduled Reboot

If switch it on, you can manage your gateway to reboot automatically as you like. There are four reboot types for you to choose, “By Day, By Week, By Month and By Running Time”.


                                            Figure 2-4-2 Reboot Types

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

Tools and Information

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

Reboot Tools

You can choose system reboot and Asterisk reboot separately.

                                      Figure 2-5-1 Reboot Prompt

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

                  Table 2-5-1 Instruction of reboots



System Reboot 

This will turn off your gateway and then turn it back on. This will drop all current calls.

Asterisk Reboot

This will restart Asterisk and drop all current calls.

Update Firmware

We offer 2 kinds of update types for you, you can choose System Update or System Online Update. System Online Update is an easier way to update your system, if you choose that, you will see some information below.

                               Figure 2-5-2Prompt Information

Upload and Backup Configuration

If you want to update your system and remain your previous configuration, you can first backup configuration, then you can upload configuration directly. That will be very convenient for you. 

                                Figure 2-5-3 Upload and Backup

Restore Configuration

Sometimes there is something wrong with your gateway that you don’t know how to solve it, mostly you will select factory reset. Then you just need to press a button, your gateway will be reset to the factory status.

                          Figure 2-5-4 Factory Reset


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 2-5-5 System Information

3. T1/E1


                                               Figure 3-1-1 General Settings

                  Table 3-1-1 Definition of General Settings


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

Interface Type

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

Table 3-1-2 Definition of advanced interface type



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

                                                  Figure 3-1-2Port Details

             Table 3-1-3 Definition of Port Details



Timing Source

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


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


Framing method for this interface


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.


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.


It shows you what signaling the port uses.

Switch Type

Only used for PRI


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


Advanced: Interface Type

                                   Figure 3-2-1 Advanced: Interface Type

        Table 3-2-1Definition of Interface Type



Echo Cancellation

Whether or not to enable echo cancellation on this line

RX Gain Whole number

-24 to 24 and multiple of 3

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

TX Gain Whole number

-24 to 24 and multiple of 3

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

ISDN: Signaling

                                                  Figure 3-2-2 ISDN: Signaling

              Table 3-2-2 Definition of Signaling



Q.SIG Channel Mapping

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

Enable Caller ID

Whether or not to use caller ID

PRI Dial Plan for Dialed Number

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

PRI Dial Plan for Dialing Number

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


Do not pass display text data.

Name_ initial:

Use display text in SETUP/CONNECT messages as the party name.

Name_ update:

Use display text in other messages (NOTIFY/FACILITY)for COLP name update.


Combined name_ initial and name_ update options.


Pas any unused display text data as an arbitrary display message during a call. Sent text goes out in default to ‘name’

Display Receive

Send/receive ISDN display IE options. The display options are a comma separated list of the following options:


Do not pass display text data.

Name_ initial:

Use display text in SETUP/CONNECT messages as the party name.

Name_ update:

Use display text in other messages (NOTIFY/FACILITY) for COLP name update.


Combined name_ initial and name_ update options.


Pas any unused display text data as an arbitrary display message during a call. Sent text goes out in default to ‘name’

Overlap Dialing

Enable overlap dialing mode--sending overlap digits.

Allow Progress When Call Released

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

Out-of-Band Indications

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

Facility-based ISDN 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.

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


Link Set Settings

                                                   Figure 3-3-1 Link Set Settings

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


                                  Figure 3-3-2 SS7 Link Set Settings

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




The linkset’s name


The linkset is enable or disable

Enabled_ st

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

Use Connect

Reply incoming call with CON rather than ACM and ANM

Hunting Policy

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


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


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


Running under SS7 standard


The point code for this SS7 signaling point


The destination point (peer) code

Set to Default

Set the linkset as the default linke set


Link Settings 

                                                 Figure 3-3-3 Link Settings

You can click button as shown below.

                                                              Figure 3-3-4 SS7 Link Settings

SS7 Config. File Backup and Restore

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


Advanced: Interface Type

                                         Figure 3-4-1 Advanced: Interface Type

          Table 3-4-1 Definition of Interface Type 



Echo Cancellation

Whether or not enable echo cancellation on this line

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

Gain for the rx (receive_ into Asterisk) channel.


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

Gain for the tx (receive_ into Asterisk) channel.


MFC/R2: Signaling 

                                        Figure 3-4-2 MFC/R2: Signaling

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



Enable Caller ID

Whether or not to use caller ID


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


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


Max amount of ANI to ask for


Max amount of DNIS to ask for


Usually national-subscriber works just fine; you can change this setting from the dialplan ; by setting the variable MFCR2-CATEGORY;(remembering ti set-MFCR2-CATEGORY from originating channels);MFCR2-CATEGORY will also be a variable in your context; on incoming calls set to the value received from the far end;mfcr2-category=national-subscriber


MFC/R2 value in milliseconds for the MF timeout


MFC/R2 value in milliseconds for the metering pulse timeout


VOIP Endpoints

SIP Endpoints

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

                                              Figure 4-1-1 SIP Status

Main Endpoint Settings

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

There are 3 kinds of registration types for choose. You can choose Anonymous, Endpoint registers with this gateway or This gateway registers with the endpoint.

You can configure as follows:

If you set up a SIP endpoint by registration “None” to a server, then you can’t register other SIP endpoints to this server. (If you add other SIP endpoints, this will cause Out-band Routes and Trunks confused.)

                                      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.

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

     Table 4-1-1 Definition of SIP Options




A name which is able to read by human. And it’s only used for user’s reference.


User name the end point use to authenticate with the gateway


Password the endpoint will use to authenticate with the gateway. Allowed characters


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.


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.

Advanced: Registration Options

          Table 4-1-2 Definition of Registration Options



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.


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


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

Qualify frequency 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.

Call Settings

                              Table 4-1-3 Definition of Call Options




Set default DTMF Mode for sending DTMF. Default: rfc2833. 
Other options: 'info', SIP INFO message (application/ dtmf-relay);
'Inband', Inband audio (require 64kbit codec - alaw, ulaw).

Trust Remote-Party-ID

Whether or not the Remote-Party-ID header should be trusted.

Send Remote-Party-ID

Whether or not to send the Remote-Party-ID header.

Caller ID Presentation

Whether or not to display Caller ID.

Advanced Timer Settings


                      Table 4-1-4 Definition of Timer Options



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 Interval

Maximum session refresh interval in seconds. Defaults to 1800s.

Session Refresher

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

Advanced: Signaling Settings

                        Table 4-1-5Definition of Signaling Options



Progress Inband

If we should generate in-band ringing. Always use ‘never’ to never use in-band signalling,

Even in cases where some buggy devices might not render it. Valid values: yes, no, never. Default: never.

Append user=phone to URI

Whether or not to add;’ user=phone’ to URIs that contain a valid phone number.

Add Q.850 Reason Headers

Whether or not to add Reason header and to use it if it is available.

Honor SDP Version

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

This option is on.

Allow Transfers

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

Allow 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 100 Trying when the endpoint registers.


                               Table 4-1-6 Definition of Fax Options




Working mode T.38 and T.30



Error Correction

Error Correction

Max Datagram

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

Fax Detect

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

Fax Activity

activate T38 fax gateway with ‘timeout’ seconds

Fax Timeout

activate T38 fax gateway with ‘timeout’ seconds

IAX2 Endpoint

                                                       Figure 4-1-5 IAX2 Endpoint

You can click button as shown below

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


               Table 4-1-6 Definition of IAX2 Endpoint




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 the endpoint will use to authenticate with gateway.

Allowed characters


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.


Authentication method for connections


Disable or not IAX2 native transfer


Use IAX2 trunking with this host


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.


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


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).


Advanced SIP Settings


                                Table 4-2-1 Definition of Networking Options



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.

Advanced: NAT Settings

                        Table 4-2-2 Definition of NAT Settings Options



Local Network

Format: or A list of IP address or IP ranges which are located inside a NATed network. This gateway will replace the internal IP address in SIP and SDP messages with the external IP address when a NAT exists between the gateway and other endpoints.

Local Network List

Local IP address list that you added.

Subscribe Network Change Event

Through the use of the test_stun_monitor module, the gateway has the ability to detect when the perceived external network address has changed. When the stun_ monitor is installed and configured, chan_sip will renew all outbound registrations when the monitor detects any sort of network change has occurred. By default this option is enabled, but only takes effect once res_stun_monitor is configured. If res_stun_monitor is enabled and you wish to not generate all outbound registrations on a network change, use the option below to disable this feature.

Match External Address Locally

Only substitute the 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= External address=

External Hostname

The external hostname (and optional TCP port) of the NAT.

External Hostname=hostname[:port] is similar to

“External address”. Examples:

External Hostname=foo.dyndns.net

Hostname Refresh Interval

How often to perform a hostname lookup. This can be useful when your NAT device lets you choose the port mapping, but the IP address is dynamic. Beware, you might suffer from service disruption when the name server resolution fails.

Advanced: RTP Settings

              Table 4-2-3 Definition of RTP Settings Options



Start of RTP Port Range

Start of range of port numbers to be used for RTP.

End of RTP port Range

End of range of port numbers to be used for RTP.

Parsing and Compatibility

                      Table 4-2-4 Instruction of Parsing and Compatibility



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.

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.


                       Table 4-2-5 Instruction of Security



Match Auth Username

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


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.


          Table 4-2-6 Instruction of Media



TOS for SIP Packets

Sets type of service for SIP packets

TOS for RTP Packets

Sets type of service for RTP packets

Codec Settings

Select codecs from the list below.


                              Figure 4-2-1 Codec Settings

Advanced IAX2 Settings

                  Table 4-3-1 Instruction of General



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 (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.


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.


You may specify a global default language for users. This can be specified also on a per-user basis. If omitted, will fallback to English(en)

Account Code

You may specify a default account for Call Detail Records (CDRs) in addition specifying on a per-user basis.

                          Table 4-3-2 Instruction of Music on Hold




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.


You may specify a global default language for users. This can be specified also on a per-user basis. If omitted, will fallback to English(en)

                          Table 4-3-3 Instruction of Codec Settings



Band Width

Specify bandwith of low, medium, or high to control which codes are used in general


Fine tune codes here using “allow” and “disallow” clause with specific codes


Fine tune codes here using “allow” and “disallow” clause with specific codes

Codec Priority

Codec priority controls the codec negotiation of an inbound IAX2 call. This option is inherited to all user entity separately which will override the setting in general.

                                Table 4-3-4 Instruction of Jitter Buffer



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


When the jitter buffer notice a significant change in delay that continue over a few frames, it will resync, assuming that the change in delay was caused by a timestamping mix-up. The threshold for noticing a change in delay is measured as twice the measured jitter plus this resync threshold.

Max Jitter Interps

The maximum number of interpolation frames the jitter buffer should return in a row. Since some clients do not send CNG/DTX frames to indicate silence, the jitter buffer will assume silence has begun after returning this many interpolations. This prevents interpolating throughout a long silence.   

Jitter Target Extra

Number of milliseconds by which the new jitter buffer will pad its size. The default is 40, so without modification, the new jitter buffer will set its size to the jitter value may help if your network normally has low jitter, but occasionally has spikes.

                         Table 4-3-5 Instruction of Misc Settings



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.


The ‘maxcallnumbers-nonvalidated’ is used to set the combined number of call numbers that can be allocated for connections where call token validation has been disabled. Unlike the ‘maxcallnumbers’ option, this limit is not separate for each individual IP address. Any connection resulting in a non-call token validated call number being allocated contributes to this limit. For use cases, see the call should be sufficient in most cases.


      Table 4-3-6 Instruction of Quality of Service




Type of service


Class of service

Advanced Fax Settings

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



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


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

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. 

                                            Figure 5-1-1 Routing Rules

You are allowed to set up new routing rule by , and after setting routing rules, move rules’ order by pulling up and down, click button to edit the routing and to delete it. Finally click the button to save what you set. shows current routing rules. Otherwise you can set up unlimited routing rules.

Call Routing Rule

There is an example for Routing rules number conversion, it transform calling, called number at the same time. Suppose you want eleven numbers start at 159 to call the eleven numbers of start at 136. Calling transform delete the three numbers from left, then writing number 086 as prefix, delete the last four numbers, and then add number 0755 at the end, it will show caller name is China Telecom. Called transform adds 086 as prefix , and Change the last two number to 88.

processing rules



Match pattern




Caller Name

Calling Transformation






China Telecom

Called transformation







                                                                   Figure 5-1-2

You can click button to set up your routings.

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

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

                     Table 5-1-1 Definition of Routing Options



Routing Name

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

Call Comes in From

The launching point of incoming calls.

Send call Through

The destination to receive the incoming calls.


                           Table 5-1-2 Description of Advanced Routing Rule



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).
X matches any digit from 0-9
  matches any digit from 1-9
matches any digit from 2-9
[1237-9]  matches any digit in the brackets (example: 1,2,3,7,8,9)
wildcard: matches one or more dialed digits.
prepend:  Digits to prepend to a successful match.
If the dialed number matches the patterns specified by the subsequent columns, then this will be prepended before sending to the trunks.

prefix:   Prefix to remove on a successful match.
The dialed number is compared to this and the subsequent columns for a match.
Upon a match, this prefix is removed from the dialed number before sending it to the trunks.

match pattern:   The dialed number will be compared against the prefix + this match pattern.
Upon a match, the match pattern portion of the dialed number will be sent to the trunks

SDfR(Stripped Digits from Right): The amount of digits to be deleted from the right end of the number. If the value of this item exceeds the length of the current number, the whole number will be deleted.

RDfR( Reserved Digits from Right) :Designated information to be added to the right end of the current number.

StA(Suffix to Add):Designated information to be added to the right end of the current number.

Caller Name: What caller name would you like to set before sending this call to the endpoint. Native language charset is allowable, e.g. Chinese charset, Latin charset.

Forward Number

What destination number will you dial? 

This is very useful when you have a transfer call.

Failover Call Through Number

The gateway will attempt to send the call out each of these in the order you specify.

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

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

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

                                                 Figure 5-1-5Change Rules

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

                                      Figure 5-1-6 Failover Call Through Number

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


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


                                           Figure 5-2-1 Establish Group

6. Network

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

WAN/LAN Settings

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

                          Figure 6-1-1 WAN/LAN Settings Interface

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




The name of network interface.


The method to get IP.

Static: manually set up your gateway IP.

DHCP: automatically get IP from your local LAN.


Physical address of your network interface.


The IP address of your gateway.


The subnet mask of your gateway.

Default Gateway

Default getaway IP address.

Basically this info is from your local network service provider, and you can fill in four DNS servers.


                       Figure 6-1-2 DNS Interface

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

DDNS Settings

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

                                  Figure 6-2-1 DDNS Interface

         Table 6-2-1 Definition of DDNS Settings




Enable/Disable DDNS(dynamic domain name server)


Set the type of DDNS server.


Your DDNS account’s login name.


Your DDNS account’s password.

Your domain

The domain to which your web server will belong.


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

                        Figure 6-3-1 Network Connectivity Checking

7. Advanced

Asterisk API

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

                   Figure 7-1-1 API Interface

               Table 7-1-1 Definition of Asterisk API




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.


If you want to deny many hosts or networks, use char & as separator. Example: or


If you want to permit many hosts or network, use char & as separator. Example: or


General information about the system and ability to run system management commands, such as Shutdown, Restart, and Reload.


Information about channels and ability to set information in a running channel.


Logging information. Read-only. (Defined but not yet used.)


Verbose information. Read-only. (Defined but not yet used.)


Permission to run CLI commands. Write-only.


Information about queues and agents and ability to add queue members to a queue.


Permission to send and receive UserEvent.


Ability to read and write configuration files.


Receive DTMF events. Read-only.


Ability to get information about the system.


Receive NewExten and Var Set events.  Read-only.


Permission to originate new calls. Write-only.


Select all or deselect all.

Once you set like the above figure, the host is allowed to access the gateway API. Please refer to the following figure to access the gateway API by putty. is the gateway’s IP, and 5038 is its API port.

                                           Figure 7-1-2 Putty Access

Asterisk CLI

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

                  Figure 7-2-1 Asterisk Command Interface

             Table 7-2-1 Definition of Asterisk CLI




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. 

                   Figure 7-3-1 Configuration Files List

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

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.


The following should be prepared before anto provisioning being applied.

l  Enable the auto provisioning in gateway

l  The ACS has been prepared

l  The network between gateway and ACS is connected

 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.

Table 7-4-1 Definition of Auto Provision




Whether to enable or disable Auto Provision

DHCP Option 66

Get ACS server address from Option 66 via DHCP


Set protocol of connection

Auto Config Server URL

The config server domain or IP address

User Name

The account of downloading from ACS


The password of downloading from ACS


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




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

 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




The firmware image


The wildcard configuration file for the whole gateway


The default(factory) configuration file


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


FW_NAME=DGW100x-current.bin    //Firmware image name

FW_MD5=b3603f3c3b5e7eb6326498640f151c79 //The md5 of firmware image

FW_VERSION=1.1.2  //Firmware version


CONFIG_NAME=defconfig.tar.gz   // default configuration file(compressed)

CONFIG_MD5KEY=2cd2dfbe52482405350816e3698cb530 // the md5 of default configuration file








NTP_SERVER1= 0.cn.pool.ntp.org

NTP_SERVER2= time.nist.gov

NTP_SERVER3= time.windows.com


















(3). Defconfig.tar.gz


Figure 7-4-2 the overview of defconfig.tar.gz

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.

       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

  1. Enable the check interval in LOGS->Log settings->System Notice like figure 7-4-5.

Figure 7-4-5 Check interval setting

  1. Configuring the ACS(Generate the md5 of firmware and defconfig.tar.gz)

l  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


The demo of E1 gateway mac address is A0:98:05:01:DB:CA (eth0), therefore the private configuration file is EPC-a0980501dbca.conf.

l  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

Figure 7-4-8 Common.conf

Figure 7-4-7 EPC- a0980501dbca.conf

l  Start TFTP service. Tftpd32.exe is a useful TFTP tools in windows7, then make sure TFTP server is select.

Figure 7-4-8 Demo TFTP server

  1. The system will receive an auto provision message in web GUI.

Figure 7-4-9 System notice logs

Figure 7-4-10 Auto provision upgrade notification

2.Restart the system. It will take about 3 minutes almost to download, upgrade Firmware and update configurations.

Figure 7-4-11Downloading the firmware and configs

Figure 7-4-12Applying the firmware and configs


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.

Parameters in SNMP setting

Table 7-5-1Definition of SNMP setting



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:


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)

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.

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.

 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.

(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, 161 and public respectively. Object ID is the node of SNMP MIBs, e.g. “.” is system location and “.” 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 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


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.

Table 7-6-1 Definition of TR069 configuration interface



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 network management server to connect back to the device.

Figure 7-6-1 TR069 configuration interface

 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



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


To specify a port that is either source or destination direction


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

8. Logs

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.

Log Settings

                                                            Figure 8-1-1 Logs Settings

                                           Figure 8-1-2 System Logs Output

                          Table 8-1-1 Definition of Logs



Auto clean:

(System Logs)

switch on : when the size of log file reaches the max size,

the system will cut a half of the file. New logs will be retained.

switch off : logs will remain, and the file size will increase gradually.

default on, default size=1MB


Asterisk console verbose message switch.


Asterisk console notice message switch.


Asterisk console warning message switch.


Asterisk console debug message switch.


Asterisk console error message switch.


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=2MB

SIP Logs:

Whether enable or disable SIP log.

Auto clean:

(SIP logs)

switch on : when the size of log file reaches the max size,

the system will cut a half of the file. New logs will be retained.

Switch off: logs will remain, and the file size will increase gradually.

default on, default size=2MB

IAX2 Logs

Whether enable or disable IAX log

Auto clean

switch on : when the size of log file reaches the max size,

the system will cut a half of the file. New logs will be retained.

Switch off: logs will remain, and the file size will increase gradually.

default on, default size=2MB

MFC/ R2 Logs

Whether enable or disable MFC/ R2 Logs log.

Auto clean

switch on : when the size of log file reaches the max size,

the system will cut a half of the file. New logs will be retained.

Switch off: logs will remain, and the file size will increase gradually.

default on, default size=2MB

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= 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.

.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=2MB

Call Statistics

Whether enable or disable Call Statistics.

System log

       System log record every time power on, power off and firmware upgrade information.


Figure 8-2-1 System Log

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

 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.

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.


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.