DINSTAR DAG3000-24O FXO Analog VoIP Gateway User Manual

: June 16, 2024
: DINSTAR

Table of Contents

DINSTAR DAG3000-24O FXO Analog VoIP Gateway
Product Information
Frequently Asked Questions
Product Introduction
Quick Installation
Basic Operation
Configurations on Web Interface
Management
Tools
References
Read User Manual Online (PDF format)
Download This Manual (PDF format)

DINSTAR DAG3000-24O FXO Analog VoIP Gateway

DINSTAR-DAG3000-24O-FXO-Analog-VoIP-Gateway-image

Product Information

Specifications

Product Name: DAG3000-24O FXO Analog VoIP Gateway
Manufacturer: Shenzhen Dinstar Co., Ltd.
Address: Floor 18, Building 7A, Vanke Cloud City Phase 1, Xingke 1st Street, Xili Sub-district, Nanshan District, Shenzhen
Postal Code: 518052
Telephone: +86 755 61919966
Fax: +86 755 2645 6659
Email: sales@dinstar.com
Website: www.dinstar.com
Document Name: DAG3000-24O FXO Analog VoIP Gateway User Manual V1.0
Document Version: V1.0

Product Introduction

Overview

The DAG3000-24O Analog Gateway for VoIP is a rich-feature FXO VoIP Gateway. It provides seamless integration between traditional analog telephony systems and IP-based communication networks.

Application Scenario

The DAG3000-24O can be used in various scenarios, including:

Small to medium-sized businesses
Call centers
Telecom operators
Service providers

Product Appearance

Appearance of DAG3000-24O

The DAG3000-24O has a compact and sleek design. It is built with high-quality materials to ensure durability and reliability.

Ports and Connectors

The DAG3000-24O features the following ports and connectors:

FXO ports: These are analog ports used to connect traditional analog telephone lines to the gateway.
RJ45 Ethernet port: This port is used to connect the gateway to the IP network.
Power connector: This connector is used to connect the gateway to a power source.

Features & Functions

The DAG3000-24O offers the following features and functions:

Support for up to 24 FXO ports
High-quality voice transmission
Advanced call routing and management capabilities
Flexible configuration options
Integration with existing telephony systems
Remote management and monitoring
Security features to protect against unauthorized access

Quick Installation

Installation Attentions

Before installing the DAG3000-24O, please take note of the following:

Ensure that you have the necessary tools and equipment for installation.
Read and understand the safety precautions mentioned in the user manual.
Choose a suitable location for the gateway, considering factors such as ventilation and accessibility.
Ensure that all cables are properly connected and secured.

RJ45 Wire Sequence

The RJ45 wire sequence for connecting the gateway to the IP network should follow the standard Ethernet wiring configuration.

Installation Steps

To install the DAG3000-24O, follow these steps:

Choose a suitable location for the gateway.
Connect the FXO ports to the analog telephone lines using appropriate cables.
Connect the RJ45 Ethernet port to the IP network using an Ethernet cable.
Connect the power connector to a power source.
Power on the gateway and wait for it to initialize.

Basic Operation

Call out and Call in via FXO Port

Call Out

To make a call using the FXO port, follow these steps:

Pick up an analog telephone connected to the FXO port.
Dial the desired phone number using the telephone keypad.
Wait for the call to connect.

Call In

To receive a call using the FXO port, follow these steps:

Wait for an incoming call on the analog telephone connected to the FXO port.
Pick up the telephone to answer the call.

Frequently Asked Questions

Q: How many FXO ports does the DAG3000-24O support?

A: The DAG3000-24O supports up to 24 FXO ports.

Q: Can I remotely manage and monitor the DAG3000-24O?

A: Yes, the DAG3000-24O provides remote management and monitoring capabilities.

DAG3000-24O FXO Analog VoIP Gateway User Manual V1.0
Shenzhen Dinstar Co., Ltd.
Address: Floor 18, Building 7A, Vanke Cloud City Phase 1, Xingke 1st Street, Xili Sub-district, Nanshan District, Shenzhen. Postal Code: 518052 Telephone: +86 755 61919966 Fax: +86 755 2645 6659 Email sales@dinstar.com Website www.dinstar.com
1

Preface

Welcome

Thanks for choosing the DAG3000-24O Analog Gateway for VoIP! We hope you will make full use of this rich-feature FXO VoIP Gateway. Contact us if you need any technical support: +86-755-61919966.

About This Manual
This manual provides information about the introduction of the analog VoIP gateway, and about how to install, configure or use it. Please read this document carefully before install the gateway.

Intended Audience
This manual is aimed primarily at the following people: Users Engineers who install, configure and maintain the gateway.

Revision Record
Document Name DAG3000-24O FXO Analog VoIP Gateway User Manual V1.0

Document Version V1.0

Firmware Version

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Quick Installation

Product Introduction

1.1 Overview
DAG3000-24O is a multi-functional analog gateway offering seamless connectivity between IP-based telephony networks and legacy telephones (POTS), and PBX systems. It adopts modularized hardware design that allows to expand FXO ports by adding boards according to user’s requirements. Each board has 8 FXO ports and the gateway supports 24 FXO ports at maximum. DAG3000-24O supports the standard SIP protocol and it’s compatible with leading IMS/NGN platforms and SIP-based IP telephony systems. Which is ideally suited for small and medium businesses, call centers and multi-location environments that need VoIP services.
1.2 Application Scenario
The application scenario of DAG3000-24O is shown as follows:

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1.3 Product Appearance
1.3.1 Appearance of DAG3000-24O
Front View:
Back View:
1.3.2 Ports and Connector

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The description of interfaces of DAG3000-24O

Port Name

Connector Description

Power Jack

Power Jack To connect 100-240V AC 50-60HZ power supply

Power Switch

Power Switch Turn on or turn off power supply

LAN Port (GE0/GE1)

RJ45

To connect to the IP network over a DSL modem or Router or a LAN switch

FXO Ports 0~7 Console Port
Reset Button

RJ45 RJ48
Reset Button

To connect to PSTN
Console port is used to carry out maintenance-related configurations
After the device is powered on, short press the RST button to restart the device; long press the RST button for about 6-10 seconds to restore the device to factory default settings.

The description of indicators of DAG3000-24O Indicator Definition Status Description

PWR

On

Power

Indicator

Off

Power supply is normal .
There is no power supply or power supply is abnormal.

RUN

Running Indicator

Slow

The device is running properly

Flashing

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Fast

SIP account is registered successfully.

Flashing

Off

The device is running improperly

Telephone On

FXO 0~7 In-use

Indicator

Off

FXO port is currently occupied by FXO port is idle or faulty

LAN Port (GE0/GE1)

Flashing Link (Green)
Off

The gateway is properly connected to network
The gateway is not connected to network or network connection is improper

1.4 Features & Functions
Key Features
Modularized Gateway with 24 FXO ports Fax over IP (T.38 and Pass-Through) Support IPv4 and IPv6 TR069 and SNMP Multiple codecs: G.711A/U,G.723.1,G.729A/B etc. Fully compatible with leading IMS/NGN, SIP based IP telephony system
Physical Interfaces
Capacity Range from 8 to 32 FXO
User Board 2 RJ45 connectors with 8 FXO
MCU Board 1RS232, 115200bps 2*10/100/1000Mbps, RJ45

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Voice Capabilities & Fax
G.711A/U law, G.723.1, G.729A/B, AMR, G.726, iLBC Silence Suppression Comfort Noise Generation(CNG) Voice Activity Detection(VAD) Echo Cancellation(G.168), with up to 128ms Adaptive (Dynamic) Jitter Buffer Programmable Gain Control T.38/Pass-through Modem/POS DTMF mode: Signal/RFC2833/INBAND VLAN 802.1P/802.1Q Layer3 QoS and DiffServ
FXO
Connector: RJ45 with 4 FXO Dial Mode: DTMF/Pulse Dialing Caller ID: FSK, DTMF Polarity Reversal Answer Delay Busy Tone Detection No Current Detection
VoIP
Protocol: SIP v2.0 (UDP/TCP),RFC3261,SDP,RTP(RFC2833), RFC3262,3263,3264,3265, 3515,2976,3311
SIP TLS/SRTP RTP/RTCP, RFC2198, 1889 RFC4028 Session Timer RFC3266 IPv6 in SDP RFC2806 TEL URI RFC3581 NAT,rport Outbound Proxy DNS SRV/ A Query/NATPR Query SIP Trunk Early Media/Early Answer

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NAT:STUN, Static/Dynamic NAT

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Software Features
Port Group Web ACL Telnet ACL Action URL Digitmap Routing Rules based Prefixes Caller/Called Number Manipulation
Network
Static/Dynamic IP, PPPoE DHCP Client IPv4/IPv6 TCP, UDP,TFTP, FTP, ARP,RARP, Ping, NTP, SNTP, HTTP/HTTPS, DNS Ping / Tracert DHCP Option 66,120,121
Environmental
1+1 Power Supply: 100-240VAC, 50-60 Hz Power Consumption:60W(Typical) Operating Temperature:0 ~ 45 Storage Temperature: -20 ~80 Humidity:10%-90% Non-Condensing Dimensions(W/D/H): 440300135mm Unit Weight: 9.3kg
Maintenance
SNMP v1/v2/v3 TR069 Auto Provisioning Web/Telnet Configuration Backup/Restore Firmware Upgrade via Web CDR

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Syslog Network Capture NTP/Daylight Saving Time IVR local Maintenance Cloud- based Management

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Quick Installation

2.1 Installation Attentions

To avoid unexpected accident or device damage, please read the following instructions before installing the DAG3000 device:
DAG3000-24O is equipped with RJ45 ports;
Anti-jamming: to reduce the interference with telephone calls, it’s highly recommended that telephone lines connected to the gateway should be placed away from power cables;
Power supply: the gateway accepts AC input voltage of 100-240V. Please ensure safe and stable power supply;
Network bandwidth: please ensure there is enough network bandwidth so as to guarantee stabilized running of the gateway
Ventilation: to avoid overheating, please do not pile up the gateway with other devices and make sure the gateway has good ventilation around.
Temperature and humidity: to avoid any accident that might cause malfunction, it’s advised to install the gateway in an equipment room where temperature and humidity are appropriate;
Mechanical load: please make sure the gateway is placed steadily to avoid damage. It’ s highly advised to horizontally place the gateway on a flat surface or a cabinet.
2.2 RJ45 Wire Sequence
DAG3000-24O is equipped with RJ45 interface as FXO port. The internal wire sequence of RJ45 cable is shown as follows:

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2.3 Installation Steps
Connect the power adapter to the power jack of the DAG3000 device, connect one end of the ground cable to the grounding port on the back of DAG3000 , and then connect the other end to the grounding bar of the shelf.
Connect network cable to the GE0/GE1 port;
Connect PSTN line to the FXO port(s).

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Basic Operation

3 Basic Operation

3.1 Call out and Call in via FXO Port

3.1.1 Call Out
One-stage Dialing: After the gateway receives a call number sent from softswitch or IPPBX, if the number matches one of the dialing rules set on Advanced Digit Map interface, the call will directly choose a FXO port to call out based on port selection rule.
Two-stage Dialing: dial a FXO port’s SIP account number from an extension of IPPBX, and then you will hear a dialing tone. After that, you will be able to dial any number of PSTN.
3.1.2 Call In
Dial the number of PSTN connected to a FXO port of the gateway, and then you will hear a dialing tone or a voice prompt of “please dial the extension number”. Then dial the called number (extension number or telephone number), after the dialing is completed, the called number will be sent to IP server such softswitch or IPPBX.
Hotline auto-dialing: Dial the number of PSTN connected to a FXO port of the gateway, then the gateway will automatically route the call to designated extension number or telephone number according to preset hotline number.

3.2 Description of Feature Codes

DAG3000-24O provides convenient telephone functions. Connect a telephone to the FXO port and dial a specific feature code, and you can query corresponding information.

Code 158# 114# 157 150
152

Description of function Dial 158# to query LAN IP Dial 114# to query the phone number of a FXO port Dial 1570 to set route mode; dial 1571 to set bride mode Dial 1501 to set IP address as static IP address; dial 1502 to set IP address as DHCP IP address Dial 152 to set IPv4 address, for example: Dial 1521921681*10# to set IPv4 address as 192.168.1.10

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156 153 160 111# 47
51# 50# 87
72
73# 90
91# 92
93# 78# 79# *200# Flash/Hook

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Dial 156 to set IPv4 gateway, for example: Dial 15619216811# to set IPv4 gateway as 192.168.1.1 Dial 153 to set IPv4 netmask, for example: Dial 15325525500# to set IPv4 netmask as 255.255.0.0 Dial 1601# to allow HTTP WAN access, Dial 1600# to deny HTTP WAN access Dial 111# to restart the device Dial 47 to allow call through IP address, for example: Dial 4719216811# to allow to call through the IP address of 192.168.1.1 Dial 51# to enable the call waiting service Dial 50# to disable the call waiting service Dial 87 to trigger blind transfer, for example: Dial 878000#, and you can blind transfer to the extension number 8000 Enable unconditional call forwarding service. Example: Dial 728000, and calls will be unconditionally forwarded to extension number 8000 Disable unconditional call forwarding service Enable the `call forwarding on busy’ service. Example: Dial 908000, and calls will be forwarded to extension number 8000 when the called number is busy Disable the call forwarding on busy’ service Enable thecall forwarding on no reply’ service. Example: Dial 928000, and calls will be forwarded to extension number 8000 when there is no reply from the called number Disable the call forwarding on no reply’ service Enable theNo Disturbing’ service Disable the `No Disturbing’ service Dial 200# to access voicemail Switch between incoming calls, If not in session, flash/hook will switch a new channel for new call.

Note

Some of the above feature codes will be realized on the available platform.

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3 Basic Operation
3.3 Send or Receive Fax
3.3.1 Fax Mode Supported
T.38IP-based Pass-ThroughVBD) Adaptive Fax Mode (automatically match with the peer fax mode)
3.3.2 Explanation of T.38 and Pass-through
T.38 is an ITU recommendation for allowing transmission of fax over IP networks in real time. Under the T.38 mode, analog fax signal is converted into digital signal and fax signal tone is restored according to the signal of peer device. Under the T.38 mode, fax traffic is carried in T.38 packages.

Pass-through: Under the pass-through mode, fax signal is not converted and fax traffic is carried in RTP packets. It uses the G.711 A or G711U codec in order to reduce the damage to fax signal.
Adaptive Fax Mode: automatically match with the fax mode of the peer device.
3.4 Local IVR Operation
3.4.1 Inquiring the IP address
Connect a PSTN line to one of the FXO ports of the gateway, and then use a mobile phone or a fixed telephone to dial the number of the PSTN line. After you hear a dialing tone or a voice prompt, dial 158# to inquire the IP address of the gateway.
3.4.2 Factory Reset
Connect a PSTN line to one of the FXO ports of the gateway, and then use a mobile phone or a fixed telephone to dial the number of the PSTN line. After hearing a dialing tone or a voice prompt, dial 166*000000#, and you will hear “successful setting”, then hang up the phone and the gateway is reset to factory defaults.

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3 Basic Operation

3.4.3 IP Address configuration
Before configuration, please ensure:
The gateway is power on; Device has been connected to network; The PSTN line has been connected to the FXO port of the gateway.

Configure dynamic IP address by DHCP: 1) Pick up the phone 2) dial 1502# and then hang up the phone 3) If the voice prompt indicates `setting successfully’, please restart the gateway after 10 seconds.

Configure Static IP address: Take the configuration of IP address `172.16.0.100′ as example.

Pick up the phone,
dial 1501# and then hang up the phone Then configure IP address and subnet mask as follow: Configure IP address
Pick up the phone 2) dial 152172160*100# and then hang up the phone. Configure subnet mask
Pick up the phone 2) dial 1532552550*0# and then hang up the phone. Configure gateway IP address
Pick up the phone 2) dial 156172160*1# and then hang up the phone. Query the IP address of the gateway:
Pick up the phone, 2) dial *158#. If the gateway uses PPPoE method to get IP addressthe IP address needs to be configures through web browser.

Note

The telephone will play voice prompt “setting successfully” if the step is correct.

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Configurations on Web Interface

4.1 Preparations for Login
4.1.1 Network Connection
Firstly, connect the device to the network and connect a PSTN line to the FXO port(See section 2.3 for details). Then dial *158# to query the IP address of the gateway.
The default IP address of the device is 192.168.11.1. It is recommended to modify the IP address of the local computer to ensure that it is in the same network segment with the device. Take windows 7 as an example, change the IP address of the local computer to 192.168.11.10:

Secondly, check the connectivity between the PC and the device. Click Start Run of PC and enter cmd to execute `ping 192.168.11.1′ to check whether the IP address of the gateway runs normally.

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4.1.2 Log In Web Interface
Open a web browser and enter the IP address of the gateway (the default IP is 192.168.11.1). Then the login GUI will be displayed.
Enter default username and password: admin/admin, then click Log in to enter into the Web interface. And then you can see the following Web interface. It is suggested that you should modify the username and password for security consideration on the Security -> Password Modification interface.

4.2 Navigation Tree
The web management system of the DAG3000-24O VoIP device consists of the navigation tree and detailed configuration interfaces.
Choose a node of the navigation tree to enter into a detailed configuration interface.

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Figure 4-1 Navigation Tree of Web Interface

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4.3 Status & Statistics
The `Status & Statistics’ menu mainly displays all kinds of information. It includes the following sub-menus: System Information, Port Status, Current Call, RTP Session, CDR, Record Statistics and VPN State .

4.3.1 System Information
Log in the Web interface, and then click Status & Statistics System Information, and the following page will be displayed. On the page, you can view the information of device ID, MAC address, network mode, IP addresses, version information, server register status and so on.

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Figure 4-3-1 System Information

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Table 4.3.1 Explanation of Items on System Information Interface:

Parameter

Explanation

Device ID

A unique ID of each device. This ID is used for warranty and cloud server authentication.

MAC address

Hardware address of the gateway There are three kinds of IP address:

DHCP: Obtain IP address automatically. DAG3000 is regarded as a DHCP client, which sends a broadcast request and looks for a DHCP server from the LAN to answer. Then the first discovered DHCP server automatically assigns an IP address to the DAG3000 from a defined range of numbers.

Static IP Address: Static IP address is a semi-permanent IP address and remains associated with a single computer over an extended period of time. This differs from a dynamic IP address, which is assigned ad hoc at the start of each session, normally changing from one session to the next.

If you choose static IP address, you need to fill in the following information:

IP Address: the IP address of the DAG3000; Subnet Mask: the netmask of the router connected the DAG3000; Default Gateway: the IP address of the router connected the
DAG3000;

IP Address

PPPoE: PPPoE is an acronym for point-to-point protocol over Ethernet, which relies on two widely accepted standards: PPP and Ethernet. PPPoE is a specification for connecting the users on an Ethernet to the Internet through a common broadband medium, such as a single DSL line, wireless device or cable modem. PPPOE IP address refers to IP address assigned through the PPPoE mode.

If you choose PPPoE, you need to fill in to fill in the following information:

DNS Server Cloud Register Status System Uptime NTP Status

Username: the account name of PPPoE Password: the password of PPPoE Server Name: the name of the server where PPPoE is placed IP addresses of primary DNS server and standby DNS server are displayed.
Whether the DAG3000 device is registered to cloud or not.
The running time of the DAG3000 device since it is powered on.
Succeed: the DAG3000 device is sync to NTP server successfully;
Failed: the DAG3000 device fails to be sync to NTP server. Then you

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Traffic Statistics
Usage of Flash Usage of RAM in Linux Usage of RAM in AOS Current Software Version
Backup Software Version
U-boot Version
Kennel version
FS Version
Hint Language

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should check network connection and the NTP server. Total bytes of message received and sent by the device. Detailed usage of Flash memory detailed RAM usage of Linux core
Detailed RAM usage of AOS The software version that runs on the DAG3000 device. Model name, version number and the software development date are displayed. Backup software is for the purpose of backup. When the current software fails, the backup software version will work. U-boot version Linux Kennel version File system version The current language of the DAG device

4.3.2 Port Status
The following figure shows the registration information of ports and port groups. Users can view the registration status of each port and port group of the gateway device through this page.
The description of ports and port groups is as follows:
Port : An FXO port
Port group: It is composed of several ports. In some cases, multiple ports can be registered with the same account and can be used for incoming and outgoing calls using the same phone number.

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4.3.3 Current Call
Call statistics for each port of the device, including: port number, port type, source, destination, connection time, and duration.
4.3.4 RTP Session
On the Status & Statistics RTP Session page, you can view the real-time RTP session information, including: port, payload type, packet period, local port, peer IP, peer port, sent packets, received packets, lost packets, jitter and duration. Figure 4-3-4 Real-time RTP Session Information

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4.3.5 CDR

CDR (Call Detail Record): is a data record produced by a telephone exchange or a telecommunication device, which contains the details of a telephone call that passes through the device.
On the Status & Statistic CDR page, you can enable the CDR function and view the details of all calls through the FXO ports of the DAG3000 device. You can also export, filter or clear the CDRs. 5000 pieces of CDRs can be saved at most.

Figure 4-3-5 CDRs of FXO Ports

4.3.6 Record Statistics
On the Status & Statistic Record Statistics page, record statistics including server status, count of current records, count of no response, count of server return errors, count of record starts, count of record startAck, count of record stops and count of stopAck are displayed.
Figure 4-3-6 Record Statistics

4.3.7 Call Limit Info
If you configure call limit on the “Call & Routing -> Call Limit” for the port, you can check the remaining call duration and number of calls of the configured port.

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4.4 Quick Setup Wizard
Quick setup wizard guides user to configure the device step by step. User only needs to configure network, SIP server and SIP port in the Quick Setup Wizard interface. Basically, after these three steps, user is able to make voice call via the DAG3000 device. For the configurations of network, SIP server and SIP port, please refer to 4.5 , 4.6 and 4.9 .

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4.5 Network
4.5.1 Local Network
The user can configure the IP protocol, IP address obtain method, management address and DNS server of the DAG3000 device on the “Network Local Network” page.
DAG3000 analog gateway supports IPv4 and IPv4&IPv6 two IP protocols and supports three IP address obtain methods (that is, automatically obtain through a DHCP server, setting a static IP address, and through PPPoE ).

Note

a. When it is configured to “obtain an IP address automatically”, it is necessary to ensure that there is a DHCP Server in the network and it is working normally.
b. The management IP address and the network IP address cannot be in the same network segment.
c. After the configuration is complete, you need to restart the device to make the configuration take effect.

Figure 4-5-1 Network Setting

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4.5.2 VLANVirtual Local Area Network
In order to control the impacts brought by broadcast storms, you can divide the local-area network into three VLAN groups, including data VLAN, voice VLAN and management VLAN on the Network VLAN page. Management VLAN transmits management-related packets, such as packets of SNMP, TR069, Web and Telnet, while voice VLAN transmits the VoIP signals and voices produced by the device itself. Data VLAN transmits data packets.
Figure 4-5-2 Configure VLAN

Table 4.5.2 Explanation of VLAN Parameters

Parameter

Explanation

Data/Voice/Management VLAN ID(0-4095)

Select what kind of messages are allowed to go through this VLAN. For example, if the checkbox on the left of data is selected, it means data messages are subject to the following network setting of this VLAN.
Set an ID to identify a VLAN based on 802.1Q protocol. Range is from 0 to 4095.

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Priority0-7 Network Setting

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Set the priority of a VLAN based on 802.1P protocol. 0 is the highest priority. Set a DHCP IP address or static IP address for a VLAN, and set the IP address of the DNS server used by the VLAN.

Note

After the configurations are finished, you need to restart the device for the configurations to take effect.

4.5.3 DHCP Option
When the DAG3000 device works as a DHCP client and applies for an IP address, DHCP server will return packets which include an IP address as well as configuration information of enabled option fields.
The following is the meaning of the option fields involved in DAG3000 (that means the following option fields are enabled, DHCP server will return information of corresponding option fields:
Option 15: to set a DNS suffix; Option 42: to specify NTP server; Option 60: to define VCI (vendor class identifier) of DAG3000 on the DHCP server; Option 66: to specify TFTP server which will assign software version to DAG3000; Option 120: to fetch SIP server address; Option 121: to obtain classless static route. DAG3000 will add these static routes to the
static route table after it fetches them from DHCP server.
Figure 4-5-3 Configure DHCP Option

Network Interface: choose which VLAN to send request to DHCP server (or to receive information from DHCP server).

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4.5.4 QoS

The DAG3000 device can label QoS priority on the IP messages it sends out, so as to resolve network delay or network congestion. Meanwhile, the device can give different QoS tags for management-related packets of Web/Telnet, voice packets and signal packets.
Figure 4-5-4 Qos

4.5.5 ARP

ARP is address resolution protocol, which helps to get the MAC address of a device through its IP address. Under TCP/IP network environment, each host is assigned with a 32-bit IP address, but MAC address needs to be known for message transmission in the physical network. In the above case, ARP can help convert IP address into MAC address.
Table 4-5-5 Explanation of Parameters for ARP

4.5.6 IPv6 Network
When IPv4&IPv6 is selected as the IP protocol in the local network, information such as IPv6 address, subnet mask, default gateway, and primary/secondary DNS servers can be configured. The IPv6 network configuration interface is shown in Figure 4-5-6:

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4.6 SIP Server
SIP server is the main component of VoIP network and is responsible for establishing all SIP calls. SIP server is also called SIP proxy server or register server. Both IPPBX and softswitch can act as the role of SIP server. Figure 4-6 Configure SIP Server Information

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Table 4.6 Explanation of Parameters for SIP Server

Parameter

Explanation

IP Protocol for SIP Stack

Choose SIP protocol stack, support IPv4 and IPv6, please choose the corresponding protocol according to the actual connected server

SIP Server Address

The IP address or domain name of the SIP server. It is provided by VoIP service provider.

SIP Server port

The service port of the SIP server. It is 5060 by default.

Registration Expires

It is used to avoid excessively frequent registrations.
When the time that is set expires, the DAG3000 device will send register request to the SIP server. The time is 300s by default.

Heartbeat

Heartbeat is used to check the connection between the DAG3000 device and SIP server.

The IP address or domain name of outbound proxy server, which Outbound Proxy Address is provided by VoIP service provider.

Outbound Proxy Port

Service port of outbound proxy server. It is 5060 by default.

Retry Interval when Registration failed

The retry interval after a registration fails. Default: 30s

Registration times per

The maximum number of registrations in a second. 0 means no

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second

limitation for registrations.

SIP Transport Type

The way of SIP-based transmission. It can be UDP, TCP, TLS or Automatic. Default: UDP.

Use Random Port

If this parameter is selected, the local port of the DAG3000 device for using SIP services is chosen by random.

The UDP/TCP port of DAG3000 device for using SIP services. SIP UDP/TCP Local Port
Default SIP UDP/TCP local is 5060.

SIP TLS Local Port

The TLS port of DAG3000 device for using SIP services. Default SIP TLS local port is 5061.

Usually, SIP server does not participate in media processing. Under SIP network, media always use end-to-end negotiating. Simple SIP server is only responsible for the establishment, maintenance and cleaning of sessions, while relatively-complex SIP server (SIP PBX) not only provides basic calling and conversational support, but also offers rich services such as Presence, Find- me and Music On Hold.
SIP server based on Linux platform, such as: OpenSERsipXecxVoSMera etc.
SIP server based on windows platform, such as :mini SipServerBrekekeVoIPswitch etc.
Carrier-grade soft switch platform, such as Cisco, Huawei, ZTE etc.

4.7 IP Profile
The device supports simultaneous registration to multiple SIP servers and making calls. Different ports can be configured with different SIP server addresses and use different voice codecs as needed. IP profiles are used to create SIP server addresses, proxy servers, dialing rules, service parameters, dialing parameters, voice codec and other parameter configuration for ports. When configuring the port, you can save the IP profile index and use it. For index configuration, refer to the “Port ” page.
When the device is only registered to one SIP server, the IP profile does not need to be configured, and the default IP profile can be used. When the device needs to register to multiple SIP servers, click the “Add” button to create a new IP profile, as shown in the figure below:

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4.8 Tel Profile
The device supports setting different values for the line parameters to each port. Different ports can be configured with different gains and fax parameters as needed. Tel profile is used to create a configuration group of line parameters, service parameters, and fax parameters for the port. During port configuration, you can save the Tel profile index and use it. For index configuration, refer to the “Port ” page.
Under normal circumstances, the Tel profile does not need to be configured, and the default Tel strategy can be used.
When you need to set different line parameters, business modes, or fax modes for different ports, you can add Tel profile through the “Add” button. As shown below:

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4.9 Port

A unique SIP account used for registration can be configured for each FXO port of DAG3000 device. Parameters of the SIP account include port number, whether to register, primary display name, primary SIP user ID, primary Authenticate ID, primary Authenticate password, off-hook auto-dial number, caller ID and so on.
Figure 4-9 Configure SIP Account for Port Registration

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Table 4-9 Explanation of Parameters Related to SIP Registration

Parameter

Explanation

Port Disable port Registration IP Profile
Tel Profile
Display Name SIP User ID
Authenticate ID Authenticate password Off-hook Auto-dial Auto-dial Delay Time

The FXO port corresponding to this account
Whether to disable port temporally
Whether to enable registration for the port
IP profile (need to be created in advance, refer to 4.6 IP profile configuration)
IP policy (need to be created in advance, refer to 4.6 IP policy configuration) Tel profile(need to be created in advance, refer to 4.7 Tel profile Configuration)
Description of SIP account. It is used to identify the SIP account.
User ID of the SIP account, which is provided by VoIP service provider (ITSP) for registration. Usually it is in the form of digits similar to phone number or an actual phone number.
SIP service subscriber’s authenticate ID used for authentication of registration. It can be identical to or different from SIP User ID.
SIP service subscriber’s authenticate ID used for authentication of registration
An extension or phone number is pre-assigned here so that the number is automatically dialed as soon as user picks up the phone
How long the auto-dial number is prolonged. If it is set as 3s, the auto-dial number is dialed after 3 seconds pass.

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Callout Limit

4 Configurations on We Maximum number of call out within the configured time

4.10 Advanced
4.10.1 Line Parameter
On the Advanced line page, you can configure FXO parameters which include for call progress tone, auto gain control, fax parameters and so on.

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Table 4.10.1 Explanation of Line Parameters

Parameter

Explanation

Call Process Tone Call Waiting Tone Auto Gain Control
Work Mode
Gain mode FAX Parameter

The signal tone standard after a phone is picked up. Choose national standards from the drop-down box. Default value is USA.
Set the duration, interval and number of repetitions for the call waiting tone
Whether to enable automatic gain control
To set the FXO ports work in both Voice and Fax mode. There are several configure options:
· Voice and FAX: to be able to make call and use FAX service
· Voice Only: allows to make call only, Fax doesn’t work if you connect a fax machine
· Fax Only: allows to make Fax call only. · POS only: allows to connect POS terminal only
IP to PSTN(RX): adjust gain value to analog phone PSTN to IP(TX): adjust gain value from analog phone The DAG3000-24Odevice supports the three fax modes: T.38 (IP-based), T.30 (Pass-Through) and Adaptive Fax Mode (automatically match with the peer fax mode).

Fax Mode
Include “a=X-fax” Attribute Include “a=fax” Attribute Include “a=X-modem” Attribute Include “a=modem” Attribute
ECM
Rate
Tone Detection by
Switch into Fax Mode

There are three fax modes: T.38, T.30(Pass-through), and Adaptive.
If this parameter is enabled, “a=X-fax” attribute will be carried in SDP
If this parameter is enabled, “a=fax” attribute will be carried in SDP
If this parameter is enabled, “a=X-modem” attribute will be carried in SDP
If this parameter is enabled, “a=modem” attribute will be carried in SDP
Whether to enable `Error Correction Mode’ECM.
The rate of sending or receiving fax, default value is 14400bps.
Fax sound is detected by caller, callee or automatically.
If this parameter is enabled, the system will switch into fax

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When Detect CNG or CED

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4.10.2 FXO Parameter
On the Advanced FXO Parameter page, you can configure FXO parameters which include incoming/outgoing call from PSTN, DC impedance, busy tone detected and so on.
Figure 4-10-2 Configure FXO Parameters

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Table 4.10.2 Explanation of FXO Parameters

Parameter

Explanation

FXO Concurrent Calls

Limit the number of concurrent FXO calls ( 0 means no limit, and the maximum number is the total number of FXO ports) which means the number of call requests received by the gateway per second. to prevent the call server from initiating a large number of calls instantly, causing traffic shocks.
It is designed to prevent the server from initiating a large number of calls at the same time and causing traffic shocks.

Configuration by FXO

When the incoming call from PSTN, you can enable or disable the FXO configuration. The FXO configuration function includes Detect CID, Send Original CID and so on.

Detect CID

When a call comes to the FXO port, FXO detects the calling number and the order of ringing. The system has two modes: first ringing and then detecting CID, first detecting CID and then ringing. The PSTN line sending CID methods usually include: sending CID before ringing, and sending CID after ringing. Therefore, when FXO detects CID, it needs to be set according to the way of PSTN line sending CID.

Send Original CID when Call from PSTN

When enabled, the caller ID of the extension will display on the PSTN side when dialing the extension.When it is not enabled, the caller ID of the extension will be display the number of the FXO port.

FXO Keep On-hook until Called Answered

After enabled, when the PSTN calls into the FXO gateway, the FXO device will go off-hook after the extension number dialed is connected. If this function is disabled, when the user dials in to the FXO port, the FXO first off-hook , and then initiates a call request to the IP.

Allow Call to SIP Server without Registration

Allow the port to initiate a call request without registering to the SIP Server. At this time, the device works in point-to-point mode.

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Called Number Preferred

When making an outgoing call, the device obtains the called number from the SIP message of the remote end. According to the content of the SIP request, the called number may be obtained from the following three fields:
P-Called-Party-ID Header
Request-line ) To header

Dial Restriction

When FXO gateway calls the PSTN, set a simultaneous dialing limit ( 0 means no restriction).

One Stage Dialing

Enabled by default, the call mode of FXO gateway means that when the FXO device makes an outgoing call, the called number obtained from the SIP message is sent to the analog end digit by digit at a time.

Add # As Ending Key

When FXO gateway makes an outgoing call, it will automatically add # after the original number as the end key to dial out together.

Off-hook Delay after On-hook

When FXO gateway calls the PSTN, the delay time for the FXO device to go off- hook after on-hook (default 1000ms).

18x Response for INVITE Choose

When the device receives the INVITE request from the remote end, it sends 180 or 183 as a temporary response code to the IP side.

183 Response after FXO Off-hook

When FXO gateway calls the PSTN, after the FXO goes off-hook, it will respond to the 183 to the IP side after a specified time (200ms by default).

Auto Dial when Detect Dial tone

When FXO gateway calls the PSTN, the FXO gateway will automatically dial out when it detects the dial tone of the PSTN.

Dial Delay after 183 Response

When FXO gateway calls the PSTN, it sends 183 messages after a specified interval to dial the PSTN.

Polarity Reversal Detected

When FXO gateway calls the PSTN, the way that FXO answers the caller is to detect the polarity reversal. After enabled, if a polarity reversal is detected, it will be reported to the caller for response. If the PSTN side cannot provide the polarity reversal detected, this function is invalid.

Delay Time after FXO Off-hook

The time for the FXO device to detect the polarity reversal and answer the caller should be less than this value. The system defaults to 10s. If the time is exceeded, the called is considered to have answered. This parameter is mostly used when there is no reverse polarity on the PSTN.

Dial Mode

FXO gateway calls the PSTN and supports 3 dialing methods:

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DTMF, Pulse, Pulse before DTMF

On-hook when Busy Tone Detected Polarity Normal Detected Current Detected

After enabling this function, the FXO gateway calls the PSTN, the FXO device hang up when: busy tone detected ,polarity normal detected and current detect .

DC Impedance

The impedance parameters when FXO gateway is connected to PBX or PSTN.

FXO Min On-hook Voltage

Minimum on-hook voltage of FXO gateway

Busy Tone Detected Cadence

The busy tone detection cadence needs to be set according to the busy tone system of the PSTN. If you do not know the busy tone standard, you can use the busy tone detection function to detect the busy tone cadence.

Cadence Count

The cadence count is used to detect the validity of the busy tone. When multiple busy tone beats are continuously detected, it is as a valid busy tone.

Delta

The error value of busy tone detection cadence

OnOff Energy Threshold

The energy threshold of busy tone from On to Off .

OffOn Energy Threshold

The energy threshold of busy tone from Off to On.

Acim

The value of AC impedance

Hybrid

The value of hybrid balance parameters

4.10.3 Media Parameter
Media parameters mainly include RTP start port, DTMF parameter, preferred Vocoder, etc.

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Figure 4-10-3 Configure Media Parameters

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Table 4.10.3 Explanation of Media Parameters

Parameter

Explanation

Use Random Port
RTP Start Port UDP Checksum Validation

If this parameter is enabled, the DAG3000 device will choose a port by random as the start port for RTP. When `Use Random Port’ is not selected, you need to configure a start port for RTP. Default RTP start port is 8000
Choose whether to enable header checksum of UDP

SRTP Method

Include Disable, AUTO and Force

RFC2833 Payload Type For an incoming call, choose local or remote RFC2833 payload Preferred (Incoming Call) type as the preferred payload type

RFC2833 Payload Type Local payload value, default value is 101

DTMF Gain

Default value is 0 DB

DTMF Send Cadence

The interval for sending DTMF signal. The default value is 200ms.

Event of Flash

If this parameter is enabled, the DAG3000 device will send

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Coder Name Payload Type Packetization Time Rate Silence Suppression Prefer Codec

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flash-hook event to remote terminal, and thus user does not need to handle it locally The device supports G.729, G.711U, G.711A, G.723, G.726-16/24/32/40. When outgoing calls are made, G.729 will be used.
Each kind of coding has a unique load value, refer to RFC3551.
The time for voice packaging
Voice data flow rate; It is defaulted by system.
Default value is `disabled’. If this parameter is enabled, VoIP transmission bandwidth can be saved, and meanwhile network congestion can be avoided.
Choose local or remote codec as the preferred codec

4.10.4 Service Parameter
Service parameters include timeout for dialing, digit map,MWI message and so on.

Table 4.10.4-1 Explanation of Service Parameters

Parameter

Explanation

Timeout for off-hook

Mainly used to define a timer that when the user is off hook an analog phone without dial any digits

Timeout for dialing

With the help of dialing timeout, you can limit the time between two digits while users are typing the digits of a number through an extension. If the timeout expires, the gateway will consider the dialing has finished and will try to send message to SIP server. Default value is 4 seconds.

Timeout for answer(Outgoing call)
Timeout for answer(Incoming call)

This parameter determines how long the caller party will wait for answer when making outgoing calls through a phone.
This parameter determines how long the phone rings when there are incoming calls

No RTP Detected

If this parameter is enabled, the situation will be detected

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Period without RTP Packet

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when there is no RTP packets received during the set time period.
The time period when there is no RTP packets received.

Table 4.10.4-2 Explanation of Service Parameters

Parameter

Explanation

IP-to-IP Call Only Accept Call from ACL (SIP server or IP Trunk)
Anonymous Call
Reject Anonymous Call
Call Confirm Tone
Howl Tone Interval After Busytone (0: No Send) Max Call Duration(0: No Limit)
Domain Query Type

If this parameter is enabled, user can dial IP address through a phone to call destination gateway.
If this parameter is enabled, the device only accepts incoming call from SIP server only. Default value is not enable’. If this parameter is enabled,anonymous’ will be included in SIP message.
If this parameter is enabled, all anonymous calls will be rejected. (default is `not disable’).
After enabling, the device will play a ringback tone even if it does not receive a 180 response.
How long does the howl tone play after the busy tone
If it exceeds the set time, the call will hang up directly (default 0 means no limit)
The way to query the domain supported: A query, SRV query, and NAPTR query

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DNS Cache

Whether to enable DNS cache

Domain Re-resolution Interval The time of domain resolution interval

Echo Cancel Tail

When there is an echo during a call, the dsp can eliminate the echo. When the echo cancellation time is longer than the echo delay, the echo can be eliminated. The default is 128 milliseconds.

Digit map is used for number dialing of calls through FXO ports of the DAG3000 device.

Table 4.10.4-3 Explanation of Parameters for Digit Map

Supported Objects
Range Range Separator Subrange Wildcard Modifiers Modifiers

Digit T DTMF
[ ] () | x . ?

0-9
Timer
A digit, a timer, or one of the symbols of A, B, C, D, #, or * One or more DTMF symbols enclosed in the [], but only one DTMF symbol can be selected One or more expressions enclosed the (), but only one can be selected Separate expressions or DTMF symbols. Two digits separated by hyphen (-) which matches any digit between and including the two digits. Matches any digit of 0 to 9
Matches 0 or more times of the preceding element
Matches 0 or 1 times of the preceding element

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4.10.5 SIP Compatibility
SIP parameters include attended transfer trigger, early media, session timer, heartbeat interval and so on. Figure 4-10-5.1 Configure SIP Parameters

Table 4.10.5-1 Explanation of SIP Parameters

Parameter

Explanation

RFC3407 Support

Whether to enable RFC3407 support. If this parameter is enabled, the device will support RFC3407 which defines the SDP capability of backward compatibility.

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“From” SIP URI Includes “user=phone”

If this parameter is enabled, user=phone’ will be contained in URI. When calls are routed to PSTN network, the called number will be got from user name. Default value isnot enable’.

INVITE with “P-Preferred-Identity” Header (RFC3325)

If this parameter is enabled, “P-Preferred-Identity” header will be added in INVITE message for anonymous call (Support RFC3325).

Value of “Refer To” refers to If this parameter is enabled, `contract header’ needs to be

“Contact”

filled in in the `refer to’ field of a SIP message.

Third Party Do Not Send 18x Response

If this parameter is enabled, the third party will not send 18x response during an attended transfer.

Send BYE when Recv REFER Response (Unattended)

If this parameter is enabled, the third party will send BYE to release session after receiving REFER during a blind transfer.

Send New REGISTER when Recv 423 Response

If this parameter is enabled, the value of `expires’ header will be automatically updated and REGISTER will be re-sent after receiving of 423 response.

CSeq Start with 1

If this parameter is enabled, the value of CSeq starts with `1′.

Forbid Invalid m=line in reINVITE

If this parameter is enabled, the device will prevent `invalid m=line’ from being carried in the SDP of re-INVITE.

SIP Message with ID Header SIP message with ID field, including MAC or SN optional

ID Header Separator

Select a separator for the MAC or SN of the ID field

Call Waiting Response Code User can choose 180 or 182 as call waiting response code

RTP Mode in SDP when Call Holding

Use send only’ orinactive’ as RTP mode during call holding.

Support Call Waiting of Huawei IPPBX

If this parameter is enabled, the device will support call waiting of Huawei IPPBX.

Accept Orphan 200 OK

If this parameter is enabled, the DAG3000 device will support different `to- tag 200 OK’ in an INVITE session.

Called Number Preferred

Choose P-Called-Party-ID header or Request-Line

Caller-ID Preferred

Choose P-Asserted-Identity header or From Header

Report SDP Whatever

If this parameter is enabled, SDP will be reported anytime

18x Response Preferred

Choose 18x Response with SDP’,Last 18x Response’ or `Local Ring Tone Only’

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Flash-hook Operation Mode Choose Mode one, Mode two or Mode three

Attended Transfer Trigger Choose On-hook’ orFlash-hook +4′

Ignore ACK

When enabled, the device will not resend the 200 OK response message even if it does not receive an ACK message after going off-hook

Figure 4-10-5.2 Configure Default SIP Parameters & Early Media

Table 4.10.5-2 Explanation of Default SIP Parameters & Early Media Parameters

Parameter

Explanation

PRACK(RFC3262)

If this parameter is enabled, the DAG3000 device supports reliable transmission of provisional response

PRACK Only for 18x with If this parameter is enabled, only PRACK will be sent when

SDP

there’s SDP in 18x response

Early Media

If this parameter is enabled, the DAG3000 device supports the receiving of Early Media.

Early Answer

If this parameter is enabled, the DAG3000 device supports early answer

Session Timer (RFC4028) Whether to enable session timer’, default value is not enable’.

Session-Expires

The interval for refreshing session; default value is 1800s.
The Session-Expires header field conveys the session interval for a SIP session.

Min-SE

The minimum interval for refreshing session; default value is 1800s.
The Min-SE header field indicates the minimum value for the session interval.

Session Refresh Method

The method to refresh session; default value is INVITE.

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Figure 4-10-5.3 Configure Timer in SIP Protocol

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Table 4.10.5-3 Explanation of Timer Parameters in SIP Protocol

Parameter

Explanation

T1
T2
T4
Max Timeout
Heartbeat Interval
Heartbeat Timeout Username of OPTION(Heartbeat) for “SIP Server” Username of OPTION(Heartbeat) for “IP TRUNK” Release all call when Heatbeat Timeout
User-Agent Header Response code when Fax Reinvite was Rejected

Value of T1 timer in SIP protocol, default is 500ms Value of T2 timer in SIP protocol, default is 4000ms Value of T4 timer in SIP protocol, default is 5000ms The max timeout of sending or receiving SIP messages, default is 32000ms The interval for sending heartbeat message, Default is 10s. The timeout for heartbeat message to be sent, default to 16s
The user ID part of OPTION SIP message in the heartbeat request for SIP server
The user ID part of OPTION SIP message in the heartbeat request for IP trunk
If enabled, when the heartbeat times out, all calls will be disconnected Customize User-Agent header field value Customize the SIP response code when fax re-invite was rejected

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4.10.6 NAT Parameter

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NAT Traversal (Network Address Translator Traversal) is a computer networking technique of establishing and maintaining Internet protocol connections across gateways that implement network address translation (NAT). NAT breaks the principle of end-to-end connectivity originally envisioned in the design of the Internet. STUNSimple Traversal of UDP over NATs is a lightweight protocol that allows applications to discover the presence and types of NATs and firewalls between them and the public Internet. It also provides the ability for applications to determine the IP addresses allocated to them by the NAT. STUN works with many existing NATs, and does not require any special behavior from them. STUN doesn’t support TCP connection and H.323.
4.10.7 Speed dial
Speed dial is a function that is available on telephones which provides an easy method of calling a telephone number by pressing fewer digits on the keypad. The tool enables one to save, organize, and have easy and quick access to regularly dialed numbers.

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4.10.8 Feature Code

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Table 4.10.8 Explanation of Feature Code

Parameter

Explanation

Inquiry LAN port IP address

Dial*158# to obtain device’s LAN port IP address

Inquiry Phone Number

Dial*114# to obtain port account

Inquiry PortGroup Number

Dial *115# to obtain port group number

Setting IP Mode

1500#, means pppmodem, 1501#, means static IP, 1502#, means obtain IP address by DHCP, 1503#, means pppoe.

Configure IP Address

152+IP, set gateway IP address

Network subnet mask configure

153+subnet mask, set gateway subnet mask

Network Gateway Configure 156+gateway IP, set gateway

Reset Basic Configuration

Dial 165000000# to restore default username/password and network configuration

Reset Factory Configuration

166000000#, reset factory

Restart Device

*111#, restart device

Call holding

During a call, dial# into call hold.Recovery the call through hook flash or #

Call by IP

Directly dial the end user IP to call

Call Waiting Activate

*51#, enable call waiting function

Call Waiting Deactivate

*50#, forbid call waiting function

Blind Transfer

If the call transfer to 801, first hook flash and then dial the 87 801#

Call Forward Unconditional

72+ phone number#, transfer the call from the phone

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Activate Call Forward Unconditional Deactivate Call Forward Busy Activate Call Forward Busy Deactivate Call Forward No Reply Activate Call Forward No Reply Deactivate Do Not Disturb Activate Do Not Disturb Deactivate
Dial Voicemail

number 73#, forbid call forward unconditional 90+ forward busy number# 91#, forbid call forward busy 92+ forward no reply number#
93#, close this function 78#, enable DND function 79#, close DND function 200#, visit voice mail box

4.10.9 System Parameter
System parameters include NTP, daylight saving time, daily reboot time, web parameter, telnet parameter and remote management. NTP (Network Time Protocolis a computer time synchronization protocol.

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Figure 4-10-9 Configure System Parameters

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Table 4.10.9 Explanation of System Parameters

Parameter

Explanation

NTP Primary NTP server address Primary NTP server port Secondary NTP server address Secondary NTP server port
SYN Interval
Time Zone
Daylight Saving Time Summary System Log The Local Network Fault Detection The Local Network Interruption Detection WEB Port SSL Port

To enable or disable NTP The IP address of primary NTP server; default IP address is us.pool.ntp.org.
The service port of primary NTP server; default port is 123.
The IP address of secondary NTP server ; Default IP address is 64.236.96.53
The service port of secondary NTP server; Default port is 123
The interval to synchronize the time of the DAG3000-24O. Default value is 3600s. The time zone of the device; Default configuration is United States central time, Chicago. Enable or disable daylight saving time Record detailed call logs Save key logs when the system is running
When the network is unavailable, report the local network failure
When the network is disconnected, report the local network interruption The web port of the device; Default port is 80 The SSL port; Default is 443

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Telnet port

4 Configurations on We Listening port of telnet service; Default port is 23

Note

After Web port and Telnet port are configured, please restart the device for the configurations to take effect.

4.11 Call & Routing
4.11.1 Port Group
When two or more FXO ports need to register with a same SIP account, you can group the ports together and then set an account for the group on the Call & Routing Port Group page.
Parameters of port group include registration, primary display name, primary SIP user id, primary authentication ID and password, secondary display name, secondary SIP user id, secondary authentication ID and password, off-hook auto dial, auto dial delay time, port select, etc.
Figure 4-211-1 Add Port Group

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Table 4.11.1 Explanation of Parameter for Port Group

Parameter

Explanation

Index Description
Display Name
SIP User ID Authenticate ID Authenticate Password Off-hook Auto-Dial Auto-dial Delay time
Port Select

The NO. of the port group; It uniquely identifies a route.
The description of the port group; it is used to identify the port group.
Display name of the port group, which will be used in SIP message, for example: INVITE sip:bob@biloxi.com SIP/2.0 Via: SIP/2.0/UDPpc33.atlanta.com;branch=z9hG4bK776asdhds Max-Forwards: 70 To: Bob <sip:bob@biloxi.com> From: Alice <sip:alice@atlanta.com>;tag=1928301774 Here Bob and Alice is the display name
User ID of this SIP account, which is provided by VoIP service provider (ITSP). It is usually in the form of digit similar to phone number or an actual phone number.
SIP service subscriber’s ID for authentication; it can be identical to or different from SIP User ID.
SIP service subscriber’s password for authentication
An extension or phone number is pre-assigned here so that the number is automatically dialed as soon as user picks up the phone
How long auto-dialing will be delayed
It specifies the policy for selecting a port for ringing in the port group Ascending: the device always selects a port from the minimum number. Cyclic ascending: the device always selects a port from a number next to the number selected last time. If the maximum number was selected last time, the next selected number is the minimum number. The sequence moves in cycles like this. Descending: the device always selects a port from the maximum number. Cyclic descending: the device always selects a port from a number next to the number selected last time. If the minimum number was selected last time, the next selected number is the maximum number. The sequence moves in cycles like this. Group ring: all ports ring at the same time

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Port

Select ports for this port group

4.11.2 IP Trunk
A peer-to-peer VoIP call occurs when two VoIP phones communicate directly over IP network without IP PBX between them. IP trunk helps establish peer-to-peer call between gateway and VoIP phones. IP trunk will be used in routing configuration.
Figure 4-11-2 Configure IP Trunk

Table 4.11.2 Explanation of IP Trunk Parameters

Parameter

Explanation

Index Description Remote Address Remote Port
Heartbeat

The No. of the IP trunk; range is from 0 to 127.
The description of the IP trunk; it is used to n identify the IP trunk.
IP address or domain name of the peer device
SIP port of the peer device Whether to enable the Heartbeat’ function for the IP trunk. Default value isnot enable’. If heartbeat is enabled, the device will send “OPTION” to the peer device.

4.11.3 Routing Parameter
Routing parameter determines a call is routed before or after manipulation.

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Figure 4-311-3 Configure Routing Parameter

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Table 4.11.3 Explanation of Routing Parameters

Parameter

Explanation

Calls from IP Calls from Analog Line

Choose calls from IP network are routed before manipulation or after manipulation.
Choose calls from analog lines are routed before manipulation or after manipulation.

4.11.4 IP Tel Routing
Calls from IP network can be routed to FXO port or port group of the DAG3000 device through IP Tel routing.
Figure 4-11-4 Add IP Tel Route

Table 4.11.4 Parameter Explanation of IP Tel Routes

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Parameter Index

Explanation

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Index of the IP Tel routing; range is from 0 to127; 0 is the highest priority.

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Description Calls from Caller Prefix
Callee Prefix Calls to

Description of the IP Tel routing; it is used to identify the IP Tel routing.
Choose calls from IP trunk or SIP server; any’ means any IP addresses. The prefix of the caller number, which helps match routing exactly. Its length is less than or equal to the caller number. For example, if caller number is 2001, the caller prefix can be 200 or 2.Any’ means the prefix matches any caller number.
The prefix of the called number, which helps match routing exactly. Its length is less than or equal to the called number. If the called number is 008675526456659, the called prefix can be 0086755 or 00.,”any” means the prefix matches any called number
Which port or port group to which calls are routed.

4.11.5 Tel IP/Tel Routing
Calls from the FXO port or port group can be routed to IP trunk or ports of SIP server/other device through Tel IP/Tel routing.
Figure 4-11-5 Add Tel IP/Tel Route

Table 4.11.5 Explanation of Tel IP/Tel Route

Parameter

Explanation

Index

The index of this Tel IP/Tel routing; range is from 0 to 127. Each index cannot be used repeatedly. Routing priority: 0 is the highest priority.

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Description Calls From Caller Prefix
Callee Prefix Calls to

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The description of this Tel IP/Tel routing; it is used to identify the routing.
Choose calls are from a port or a port group
The prefix of the caller number, which helps match routing exactly. Its length is less than or equal to the caller number. For example, if caller number is 2001, the caller prefix can be 200 or 2. any’ means the prefix matches any caller number. The prefix of the called number, which helps match routing exactly. Its length is less than or equal to the called number. If the called number is 008675526456659, the called prefix can be 0086755 or 00.any’ means the prefix matches any called number.
Choose calls are routed to a port, port group, IP trunk or SIP server

4.11.6 Call Limit

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Table 4.11.6 Explanation of call limit

Parameter

Explanation

Index

The index of call limit

Description

The description of this call limit; it is used to identify the limiting.

Daily Duration

The maximum duration of a daily call

Month Duration The maximum duration of a monthly call

Daily Calls

The times of daily calls

Minute Calls

The times of a minute calls

Daily Connected The times of daily connected calls

The times of mi connected calls Minute Connected
The times of calls made in minute

Dest Port

Select the port that needs to be call limit

Note

0 means no limit 2) The call limit only affects the outgoing call from the FXO port 3) The day/month limit will be automatically reset when the NTP time
synchronization is successful.

4.12 Manipulation
Number manipulation refers to the change of a called number or a caller number during calling process when the called number or the caller number matches the preset rules.
4.12.1 IP Tel Called
On the IP Tel Called page, you can set rules for manipulating the called number of IP Tel calls.

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Figure 4-12-1 Add IP Tel Called Number Manipulation

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Table 4.12.1 Explanation of Parameters for IP Tel Called Number Manipulation

Parameter

Explanation

Index Description Calls From
Caller Prefix
Called Prefix
Calls to Stripped Digits from Left

The index of this manipulation; range is from 0 to 127. Each index cannot be used repeatedly. 0 is the highest priority
Description of this manipulation; it is used to identify this manipulation.
Determine the calls come from IP trunk or SIP server
Set a prefix for caller number. The prefix’s length is less than or equal to that of the caller number, which helps to match the caller number of this call. If caller number is 2001, the caller prefix can be 200 or 2. “any” means match any caller number.
Set a prefix for called number. The prefix’s length is less than or equal to called number, which helps to match the called number. If called number is 008675526456659, the called prefix can be 0086755 or 00. “any” means match any called number.
Determine the call is routed to a port or a port group.
The number of digits which are lessened from the left of the called number

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Stripped Digits from Right Prefix to Add
Suffix to Add

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The number of digits which are lessened from the right of the called number The prefix added to the called number after its digits are lessened. The suffix added to the called number after its digits are lessened.

4.12.2 Tel IP/Tel Caller
On the Tel IP/Tel Caller page, you can set rules for manipulating the caller number of Tel IP/Tel calls.
Figure 4-12-2 Add Tel IP/Tel Caller Number Manipulation

Table 4.12.2 Explanation of Parameters for IP Tel Called Number Manipulation

Parameter

Explanation

Index Description Calls From

The index of this manipulation; range is from 0 to 127. Each index cannot be used repeatedly. 0 is the highest priority Description of this manipulation; it is used to identify this manipulation.
Determine the calls come from a port or a port group.

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Caller Prefix
Called Prefix
Calls to Stripped Digits from Left Stripped Digits from Right Prefix to Add Suffix to Add

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Set a prefix for caller number. The prefix’s length is less than or equal to that of the caller number, which helps to match the caller number of this call. If caller number is 2001, the caller prefix can be 200 or 2. any’ means match any caller number. Set a prefix for called number. The prefix’s length is less than or equal to called number, which helps to match the called number. If called number is 008675526456659, the called prefix can be 0086755 or 00.any’ means match any called number.
Determine the call is routed to a port, a port group, an IP trunk or a SIP server.
The number of digits which are lessened from the left of the caller number
The number of digits which are lessened from the right of the caller number
The prefix added to the caller number after its digits are lessened.
The suffix added to the caller number after its digits are lessened.

4.12.3 Tel IP/Tel Called
On the Tel IP/Tel Called page, you can set rules for manipulating the called number of Tel IP/Tel calls.
Figure 4-12-3 Add Tel IP/Tel Called Number Manipulation

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Table 4.12.3 Explanation of Parameters for Tel IP/Tel Callee Number Manipulation

Parameter

Explanation

Index Description Calls From
Caller Prefix
Callee Prefix
Calls to Stripped Digits from Left Stripped Digits from Right Prefix to Add Suffix to Add

The index of this manipulation; range is from 0 to 127. Each index cannot be used repeatedly. 0 is the highest priority
Description of this manipulation; it is used to identify this manipulation.
Determine the calls come from a port or a port group.
Set a prefix for caller number. The prefix’s length is less than or equal to that of the caller number, which helps to match the caller number of this call. If caller number is 2001, the caller prefix can be 200 or 2. any’ means match any caller number. Set a prefix for called number. The prefix’s length is less than or equal to called number, which helps to match the called number. If called number is 008675526456659, the called prefix can be 0086755 or 00.any’ means match any called number.
Determine the call is routed to a port, a port group, an IP trunk or a SIP server.
The number of digits which are lessened from the left of the called number
The number of digits which are lessened from the right of the called number
The prefix added to the called number after its digits are lessened.
The suffix added to the called number after its digits are lessened.

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Management

4.13.1 TR069
TR069 is short for Technical Report 069, which provides a commonly-used framework and protocol for next-generation network devices. As an application- level protocol on top of IP TR069 has no limitation to access ways of network devices.
Under the network management model of TR069, ACS (Auto-Configuration Server) works as a management server, responsible for managing CPEs (Custom Premise Equipment).
ACS URL (auto-configuration server URL address) is provided by service provider. The ACS URL generally starts with http:// or https://
Username and password are used for ACS authentication.
Figure 4-13-1 Configure TR069 Parameter

Table 4.13.1 Explanation of TR069 Parameters

Parameter

Explanation

TR069 ACS URL Username(ACS) Password(ACS) Periodic Inform
Periodic Inform

Choose whether to enable TR069; it is not enable’ by default. The IP address or domain name of ACS, which is provided by service provider. Username of ACS, which is provided by service provider. Password of ACS, which is provided by service provider. Choose whether to enablePeriodic Inform’; if it is enabled, ACS will connect to CPE every 30 seconds (if the interval is set as 30 seconds). The interval set for periodic connection between ACS and CPE.

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Interval Username (CPE) Password (CPE) Port

Username of CPE Password of CPE The port to connect CPE and ACS

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4.13.2 SNMP
SNMP (Simple Network Management Protocol) is an Internet-standard protocol for collecting and organizing information about managed devices on IP networks and for modifying that information to change device behavior. Devices that typically support SNMP include routers, switches, servers, workstations, printers, modem racks and more.
SNMP is widely used in network management for network monitoring. SNMP exposes management data in the form of variables on the managed systems organized in a management information base which describe the system status and configuration. These variables can then be remotely queried (and, in some circumstances, manipulated) by managing applications.
Three significant versions of SNMP have been released. SNMPv1 is the original version of the protocol. More recent versions, SNMPv2c and SNMPv3, feature improvements in performance, flexibility and security.
Figure 4-13-2 Configure SNMP Parameters

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Table 4.13.2 Explanation of SNMP Parameters

Parameter

Explanation

SNMP
Community Configuration
Group Configuration

The DAG3000 device supports three versions of SNMP, namelyV1V2C and V3.
Community configuration exists in V1 and V2C. Community: fill in a community name used to read through SNMP protocol; it is a character string. Source: The IP address of SNMP server. SNMP server cannot identify the packets sent from DAG3000 unless the community configured in DAG3000 matches with the community configured in SNMP server.
Group configuration exists in V1 and V2C and V3. Group: fill in a group name which is used to identify the group; it’s a character string. Community: fill in a community which means this community has joined in the group. In the following, access permission of read, write and notify is configured for each group.

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View Configuration

View configuration exists in V1, V2C and V3.
ViewName: fill in a view name which is used to identify this view.
ViewType: choose Included’ orExcluded’. Included’ means the view includes the OID of the corresponding ViewSubtree, whileExcluded’ means the OID of the corresponding ViewSubtree is excluded from this view.
ViewSubtree: fill in the OID of the view subtree.
ViewMask: it is used to withdraw a row of a table, such as an Ethernet port.

Access Configuration

Access configuration exists in V1, V2C and V3, under which permission of read, write or notify is configured for a community group. Group: choose a group name that has been configured. Read: Choose a read’ view for the group. Write: Choose awrite’ view for the group. Notify: Choose a `notify’ view for the group.

Trap Configuration

Trap configuration exists in V1, V2C and V3, which is aimed to send trap alarm.
Trap Type: Choose V1, V2C and Inform.
Trap IP: the IP address of the destination SNMP server where trap alarm is sent.
Trap Port: the port of the destination SNMP server, which will receive trap alarm.
Trap Community: the community configured in the destination SNMP server.

User Configuration

User configuration exists in V3. When V3 transmits SNMP packets in an encryption way, this item needs to be configured. User: fill in a user name used to authenticate. AuthType: choose MD5 or SHA as authentication type. AuthPassword: the password used to authenticate. Privacy Type: Choose DES, AES or AES 128 as encryption type. Privacy Password: the encryption password.

4.13.3 Syslog
Syslog is a standard for message logging. It allows separation of the software that generates messages, the system that stores messages, and the software that reports and analyzes messages. It also provides a means to notify administrators of problems or performance.
Syslog levels include: EMERG, ALERT, CRIT, ERROR, WARNING, NOTICE, INFO and DEBUG.

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Figure 4-13-3 Configure Syslog Parameters

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When the DAG3000 device registers to SIM Cloud server, local syslog will be changed to nonconfigurable and all logs will be stored on the Cloud server.
4.13.4 Provision
Provision is used to make the DAG3000 device automatically upgrade with the latest firmware stored on an http server, an ftp server or a tftp server. Please refer to the Instruction for Using Provision.

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Figure 4-13-4 Provision

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Table 4.13.4 Explanation of Provision Parameters

Parameter

Explanation

URL Check Interval Account Password

URL of provisioning server, support HTTP, TFTP, FTP
The interval to check whether there is new firmware version on the provisioning server Account for logging in provisioning server Password for logging in provisioning server

4.13.5 Cloud server
You can register the DAG3000 device to cloud server, and then the device can be managed by the cloud server.
Figure 4-13-5 Configure Cloud Server

Table 4.13.5 Explanation of Parameters for Cloud Server

Parameter

Explanation

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Server Address

The IP address of the cloud server

Port

The listening port of the cloud server

Domain

The domain name of the cloud server

Join the remote

Choose whether to join the remote management system of the cloud

management system server.

4.13.6 User Manage
On the Management User Manage page, the administrator of the DAG3000 device can classify users in different groups, and set login username and password for each user.
Figure 4-13-6 Modify Username and Password

4.13.7 Remote Server
In case that you need remote technical support, technical support engineers can connect your device with a service server on the Management Remote Server page, so as to better help you to solve problems.
Figure 4-13-7 Configure Remote Server

4.13.8 Record Parameter
After configuring the record server, you can upload the call voice of the FXO port of the device to the record server. The record configuration page is shown in Figure 4-13-8:

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Table 4.13.8 Explanation of Record Parameters

Parameter

Explanation

RCD Server Address Rcd Port Rcd Period Select Rcd Directly To Server

Enable or disable the record function Set the record server address, IP address or domain name Recording server port (The default is 2999) Only record within the set time range, support 3 recording periods In the NAT environment, the recording can be directly sent to the public network

4.13.9 Radius Parameter
After the Radius server is configured, you can log into the gateway after successful Radius authentication. The Radius server configuration page is shown in Figure 4-13-9:

Table 4.13.9 Explanation of Parameters for Radius Config

Parameter

Explanation

Radius

Enable or disable Radius

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Local Port
Device Behavior Upon RADIUS Timeout
Server IP Server Auth Port Server Key

The port of the local Radius client
Processing after Radius authentication timeout. Verify Access Locally : After the timeout, when verifying the user name and password of the local Web login is successful, the login is successful; Deny Access : No matter what, the login is refused
The IP address of the Radius server
The port of the Radius server
The key of the Radius server

4.13.10 Action URL
Action URL is a means of allowing VoIP platform/VoIP server to learn about the statuses of the DAG3000 device. This is realized by GET request over the HTTP protocol. During the transmission of status, some data (such as device ID, mac address, called/caller number, IP address) carried in GET request can also be reported to VoIP platform/VoIP server. The data that can be carried in GET request, please refer to the notes on the Management Action URL page.
Figure 4-13-10 Configure Action URL

Event: Statuses of DAG3000 device, which will be reported to VoIP platform/VoIP server.
Action URL: for example, http://host:port/file.php?macaddr=$mac, among which host’ means the HTTP server’s IP address or domain name,port’ means the http server’s listening port, file.php’ means the script that will process this request, and$mac’ means the parameter carried in the request when this request is sent out.
Heartbeat: heartbeat packets are sent to URL by the DAG3000 device, used to examine the connection between the DAG3000 device and HTTP/HTTP server.

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4.13.11 SIP PNP
The gateway can restore the profile and upgrade the software version through SIP PNP. The SIP PNP process is as follows:

The gateway sends request packet for SIP subscription to the multicast at intervals 2) The gateway receives the Notify message and reads the URL address of the
deployment server in the message 3) Initiate the Provision to the URL to restore the profile or upgrade the software
version Figure 4-13-11 Configure SIP PNP

Table 4.13.11 Explanation of Parameters for SIP PNP

Parameter

Explanation

PNP Enable Server Address Server Port Update Interval

Enable or disable PNP
SIP PNP server IP address, and the default is the multicast address 224.0.1.75 SIP PNP server port, and the default is 5060
Send subscription messages periodically, and 3600s by default

4.13.12 NMS Configuration
NMS is a public/private cloud-based device management platform. After users register with the NMS, you can perform batch upgrades, status monitoring, and view alarm information for the devices. To apply for cloud management server(NMS) , please contact your local distributor.

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4.14 Security
4.14.1 WEB ACL
ACL (Access Control List) for Web is used to configure IP addresses that are allowed to access the Web Interface of the DAG3000 device. The IP address list can’t be null once ACL is enabled. Figure 4-14-1 Add IP Address to Web ACL

4.14.2 Telnet ACL
ACL (Access Control List) for Telnet is used to configure IP addresses that are allowed to access the Telnet Interface of the DAG3000 device. The IP address list can’t be null once ACL is enabled.

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Figure 4-14-2Add IP Address to Telnet ACL

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4.14.3 Passwords
You can configure or modify the username and password for logging in the Web interface and the Telnet interface of the DAG3000 device on this page.
NoteBoth the username and password of Web and Telnet are admin’ andadmin’ by default. It is advised to modify them for security consideration.
Figure 4-14-3 Modify Username and Password

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Tools

4.15.1 Firmware Upload
On the Tools Firmware Upload page, you can upload a new firmware version from a local folder. Figure 4-15-1 Upload Firmware
Steps of Firmware Uploading: Step 1. Check the current firmware version on the Status & Statistics System Information page. Step 2. Prepare firmware package. Step 3.Upload firmware, select the package from a specific folder on the computer and click the Upload button. Step 4. Keep waiting until it prompts `Software loaded successfully!’ Step 5. Reboot the device on the Tools Device Restart page.
4.15.2 Data Backup
On the Tools Data Backup page, you can download and backup configuration data, device status and summary messages on local computer. Figure 4-15-2 Backup Data

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4.15.3 Data Restore
On the Tools Data Restore page, you can restore configuration data through uploading a data file from local computer. The restored configurations will take effect after the device is restarted. Figure 4-15-3 Restore Data
4.15.4 FXO Test
FXO test consists of two parts: Impedance Test and Auto-detect Busy Tone. Impedance Test The impedance test of FXO port is mainly for the technicians to perform impedance matching, and can only test the ports in the online state. Figure 4-15-4.1 Impedance Test

Table 4.15.4-1 Explanation of Parameters for Impedance Test

Parameter

Explanation

Test Type Port

Choose a type to test Choose a port to test

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Dial Timeout Time Acim
Hybrid
Match Mode Dial Test Number

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Set the dialing timeout time. If you are not sure, you can also perform a “Dial Test” first (see step 2 for details)
Display the current impedance value of the FXO port (displayed value, cannot be modified)
Display the current hybrid parameters of the FXO port (displayed value, cannot be modified)
Matching mode: Simple, Standard and Exact (The higher the mode, the higher the accuracy and the longer it takes).
Fill in the test number

Steps of impedance test:

Navigate to Tools> FXO Test> Impedance Test
Fill in the dial timeout time (if you don’t know the dial timeout time, you can perform the dial timeout test first (about 10 seconds), after selecting the online port to be tested, click “Dial test”, and the timeout time will be displayed after the test is completed)
Select the matching mode, test port, and test number, etc., and click “Start” (different modes, time and accuracy are also different, the simple mode is about 15 minutes, the standard is about 30 minutes, and the exact is about 45 minutes);
After the test is completed, the Acim and Hybrid values will be displayed.

Note

The dial test number can be configured by itself, but it cannot be the same as the service number.
If you do not click to save the result, after restarting, the dialing timeout time, dialing test number and impedance value will be invalid.
Please do not leave this page before the test is completed to avoid errors.

Auto-detect Busy Tone Busy tone detection can only select the online port . The testing steps are as follows: Figure 4-15-4.2 Auto-detect Busy Tone

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Table 4.15.4-2 Explanation of Parameters for Auto-detect Busy Tone

Parameter

Explanation

Test Type Port Test Number Original Cadence Recommended Cadence

Choose a type to test Choose a port to test The destination number for busy tone detection (see step 2 for details) The original busy tone cadence captured during the detection Recommended busy tone cadence after detection

Steps of Auto-detect Busy Tone:

Navigate to Tools > FXO Test > Auto-detect Busy Tone
Select the online port to be tested and fill in the test number (Make sure that the busy tone service has opened for this number. Its advised to use a PSTN line to connect telephone for test. If this parameter is null, it means no number is dialed)
Click `Start’, it will take about 1 minute, please do not leave this page
After the test is completed, the original cadence and recommended cadence are displayed, Please save the result after finishing, otherwise you can clear the results and retest.

4.15.5 Ping Test
Ping is used to examine whether a network works normally through sending test packets and calculating response time.
Instructions for using Ping:

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1. Enter the IP address or domain name of a network, a website or a device in the input box of Ping, and then click Start.
2. If related messages are received, it means the network works normally; otherwise, the network is not connected or is connected faultily. Figure 4-15-5 Execute Ping Test

4.15.6 Tracert Test
Tracert is short for traceroute, used to track a route from one IP address to another. Instruction for using Traceroute: 1) Enter the IP address or domain name of a destination device in the input box of
Traceroute, and then click Start.
Figure 4-15-6 Execute Tracert Test

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Destination: the IP address or domain name of a destination device that needs to be tracked. Max Hops: the maximum hops for searching the above IP address or domain name. For example, if `max hops’ is set as 30, and the configured IP address or domain name cannot be reached within 30 hops, it’s thought that the IP address or domain name cannot be searched. 2) View the route information from the returned message.
4.15.7 Network Capture
Network capture is an important diagnostics tool for maintenance. It is used to capture data packages of the available network ports. PCM Capture: PCM capture helps to analysis voice stream between analog phone and DSP chipset. Figure 4-15-7.1 Capture PCM Packages

Click “Start’ to enable PCM capture Dialing out through the device, start talking a short while then hang up the call. Click Stop’ to disable network capture Save the file to local computer The captured package is named capture(x).pcap’, among which x is the serial number of the capturing and will be added 1 in next time. The sample of PCM capture as below:

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Syslog Capture:
Syslog capture is another way to obtain syslog which is the same as remote syslog server and filelog. The captured file is saved as pcap format so that it can be opened in some of capturing software like Wireshark, Ethereal software etc. Figure 4-15-7.2 Capture Syslog Packages
Click “Start’ to enable syslog capture Dialing out through the device, start talking a short while then hang up the call. Click Stop’ to disable syslog capture Save the capture to local computer The capture package is named capture(x).pcap’, among which x is the serial number of capturing and will be added 1 in next time. The sample of syslog capture as below:

DSP Capture:
DSP capture helps to analyze voice stream inside DSP chipset. The DSP chipset will handle RTP from IP network as well as voice stream from analog phone.

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Figure 4-15-7.3 Capture DSP Packages

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Click Start to enable DSP capture Dialing out through the device, start talking a short while then hang up the call. Click Stop to disable DSP capture Save the capture to local computer The captured package is named `capture(x).pcap’, among which x is the serial number of the capturing and will be added 1 in next time. The sample of RTP capture as below:

4.15.8 Factory Reset
Click `Apply’ to restore configurations of the device to the factory default settings. Figure 4-15-8 Reset Device to Factory Default Setting

4.15.9 Device Restart
For some configurations or changes to the DAG3000 device, you are required to restart the device for the configurations or changes to take effect.

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Figure 4-15-9 Restart Device

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Abbr. ACD DNS SIP TCP UDP RTP POE VLAN ARP CID DND DTMF NTP DMZ STUN PSTN

5 Glossary
5 Glossary
Full Name Automatic Call Distribution Domain Name System Session Initiation Protocol Transmission Control Protocol User Datagram Protocol Real Time Protocol point-to-point protocol over Ethernet Virtual Local Area Network Address Resolution Protocol Caller Identity Do NOT Disturb Dual Tone Multi Frequency Network Time Protocol Demilitarized Zone Simple Traversal of UDP over NAT Public Switched Telephone Network

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