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DWG2000-1G User Manual v1.0
Dinstar Technologies Co., Ltd.
Address: Floor 6, Guoxing Building, Changxing Road, Nanshan District, Shenzhen, China
Postal Code: 518057
Telephone: +86 755 2645 6664
Fax: +86 755 2645 6659
Emails: [email protected], [email protected]
Website: www.dinstar.com www.dinstar.cn
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DWG2000-1G User Manual
Revision Records
Document version
Firmware version
Revised by
Date
Changes
1.0
9.50.41
Technical Support Department
2011/9/8
The first version
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DWG2000-1G User Manual
Table of Contents
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1. Equipment Introduction
This chapter mainly introduces functions and structures of DWG2000-1G.
1.1 Introduction
DWG2000-1G is full functions VoIP gateway based on IP and GSM wireless network, which provides a flexible network configuration, powerful features, and good voice quality. It works for carrier grade, enterprise, SOHO, residential users for cost-effective solution.
1.2 Scenario of Applications of Products
DWG2000-1G provides access of GSM network.
With the development of users and telecom service, mobile network and fixed network integration will be steadily increasing. DWG2000-1G provides high quality VoIP service which perfectly meets the requirement. This is a scenario shown as figure 1-2-1
Figure 1-2-1 Network scenario
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1.3 Product Appearance
The appearance of DWG2000-1G shows as follow
Figure 1-3-1 Front & Rear view of DWG2000-1G
1.3.1 Interface
Interface
Table 1-3-1 Description of interface
Description
WAN
LAN
SIM
Indications blinking when connect successful
Indications blinking when connect successful
1 SIM channel
DC DC 12V
1.3.2 Indications
Table 1-3-2 Description of indicators on front view of DWG2000-1G
Indicators Color Name Status Description
RUN
GSM
Green
Green
Register indicator
Running indicator
Off Unregistered
Slow blinking Registered
Fast Blinking SIM cards Registered
Slow blinking Unregistered
1.3.3 Reset
Push RST button for 6 seconds of DWG2000-1G, the device will restore factory setting.
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1.4 Functions and Features
1.4.1 Protocol Standard Supported
Standard SIP and MGCP(option) protocol;
Simple Traversal of UDP over NATs (STUN);
Point-to-point protocol over Ethernet (PPPoE);
Hypertext Transfer Protocol (HTTP);
Dynamic Host Configuration Protocol (DHCP);
Domain Name System (DNS);
GSM;
ITU-T G.711α-Law/μ-Law、G.723.1、G.729AB;
1.4.2 System Function
PLC: Packet loss concealment
VAD: Voice activity detection
CNG: Comfort Noise Generation
Local/Remote SIM card work mode
Adjustable gain of port
DTMF adjustment
Balance alarm
Lock/unlock SIM/UIM
Mobile number display rejection
Sending/receiving SMS
Customize IVR Recording
White and black list
One number access
Open API for SMS, support USSD
Echo Cancellation (with ITU-T G.168/165 standard)
Automatic negotiate network
Hotline
1.4.3 Industrial Standards Supported
Stationary use environment: EN 300 019: Class 3.1
Storage environment: EN 300 019: Class 1.2
Transportation environment: EN 300 019: Class 2.3
Acoustic noise: EN 300 753
CE EMC directive 2004/108/EC
EN55022: 2006+A1:2007
EN61000-3-2: 2006,
EN61000-3-3: 1995+A1: 2001+A2: 2005
EN55024: 1998+A1: 2001+A2: 2003
Certifications: FCC, CE
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1.4.4 General Hardware Specification
Power Supply:AC100~240V 50/60HZ DC12V/1A
Temperature: 0~40 ℃(Operation), -20~80 ℃(storage)
Humidity: 5%~90%RH,
Power Consumption: 5W
Dimensions: 112(W) x76(D) x24(H) mm
Net weight: 0.7kg
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2. Equipment Installation
This chapter mainly introduces DWG2000-1G hardware installation and connection of equipment.
2.1 Installation Notice
DWG2000-1G uses DC12V power. Power supply should ensure the reliability and stability,otherwise, it may damage the SIM card or device. In addition, make sure the power supply connects to ground bar well. With right ground protect connection, that can reduce the surge voltage caused by lightning that damage the equipment, and ensure voice quality (note: when calls with irregular noise occurring, please check the power whether connect ground well).
Common measures are as follows:
Making sure that all devices powered in the buildings are in accordance with NEC (National
Electric Code, National Electrical Regulations) Article 250 of manual properly grounded;
Making sure that the panel of building power supply units used high-quality copper wire well connect with the ground wire, copper wire specifications shall comply with NEC Table 250-94/95 relevant provisions of the manual. Grounding cable that buried in the building field, including at least one or several 2.44m deep under the ground, or buried deeply underground at least 0.76m, with a wire around the building (see NEC manual specifications the relevant provisions of the table 250-94/95);
Setting up voltage protector between equipment and ground connected to some other computer equipments (either directly or through other devices), such as terminal or printer must also be plugged into the same surge protector.
Network interface of DWG2000-1G supports RJ45 standard with 10Mbps or 100Mbps network.
Wireless section, inserting SIM card directly, GSM channel should work properly.
2.2 Installation Procedure
The outlook of DWG2000-1G looks like a 1U chassis; to install hardware the cable is needed.
After unpacking the equipment, please do follow the procedure as following steps:
2.2.1 Install SIM Card
When installing SIM card, opening blank panel of SIM slot, procedure shows as below:
Push down the yellow button,the SIM slot will popup;
Inset the SIM card to the SIM slot.
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2.2.2 Antenna Installation
Take antenna connected in antenna interface of DWG which sign of "ANT" on
2.2.3 Cable Connection of Equipment
DWG2000-1G works in Route mode
Figure 2-2-1 DWG2000-1G connection
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3. Network Configuration
In this chapter we will introduce the initial configuration of DWG2000-1G gateway. All of the network parameters of the gateway can be configured by IVR guidance.
3.1 Preparation
Please ensure the following stepes are done properly before IVR setting:
Prepare an analog telephone or mobile phone
Make sure the gateway is power on
Make sure the gateway is connected with the network
Completed the SIM installation
Make sure that the current mobile network is working
3.2 Attentions
In each step, if user hears an IVR message of “setting successful”, which means that user has finished this step successfully. However, if user hears a “setting failed” message, please check redo theat step again.
DWG2000-1G can work in two modes: route mode and bridge mode. when the gateway is under bridging mode, user should configure network parameters of WAN port; when the gateway is under the route mode, user should configure LAN port.
3.3 General Feature Codes for System Setting
Table 3-3-1 Feature codes for system setting
Dial numbers Features
*114# Play the phone NO.
*115#
*150*a#
Check the TT number of gateway (using just when the device interconnects
*152*a*b*c*d
Configure subnet mask. a, b, c, d are the four fields of the subnet mask
Configure the device gateway, a, b, c, d are the four fields of the device
*157
*195#
Setting the work mode (route or bridge), * 157 * 0 # is route mode, * 157 *
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*198#
*199#
*111#
Clear record
Setting Record. dial*199# start record(≤ 20s), then press # end the
3.4 Static IP Configuration
This chapter introduces IP configuration of DWG2000-1G through calling IVR.
Assuming the IP address of a DWG2000-1G device is 172.16.0.100, subnet mask is 255.255.0.0,
IP of gateway is 172.16.0.1, configured as follows:
Insert a SIM card into the DWG2000-1G gateway
1)
The configuration mode: Dial the phone number of this SIM card. hear a message, then enter
“*150*1#”, hang up when hear “ setting successful" message;
2)
Configure IP address: Dial the phone number of this SIM card, hear a message, enter "* 152 *
172 * 16 * 0 * 100 #" hang up when hear “setting successful" message;
3)
configure subnet mask: Dial the SIM card phone number, enter "*153*255*255*0*0#" hang up when hear “setting successful" message;
4)
Configure gateway: Dial the SIM card phone number, enter "*156*172*16*0*1#" hang up when hear “setting successful" message;
5)
Please wait about ten seconds when finishing the operations, restart device. dial the SIM card phone number, enter “*158#”to check the Static IP address;
3.5 DHCP Configuration
DHCP mode configure as follows:
Insert a SIM card into a slot, dial the SIM card number. When hearing a hint message, then enter “*150*1#”, if hearing “ setting successful" message, which means the DHCP is confirued successfully;
Restart the device, wait for 30 seconds, and then dial the SIM card telephone number, enter
"* 158 #" to query the IP address;
Note: If reporting the IP address is 0.0.0.0, which means that the gateway could not obtain a IP address successfully. Please check:
Make sure the device have been connected to the network;
Make sure the DHCP Server is working. If there is no DHCP Server, please set the IP of device to static IP .
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4. WEB configuration
This charpter describes web configuration of DWG2000-1G.
4.1 Preparing
WEB configuration includes the following components: network configuration, system information, mobile configuration and system configuration.
4.2 Access the System Through HTTP
Enter IP address of DWG2000-1G in browser. The default IP of LAN port is 192.168.11.1. and the
GUI shows as below:
Figure 4-2-1 WEB log interface
Enter username and password and then click “OK” in configuration interface. The default username and password are “admin/admin”. It is strongly recommended, change the default password to a new password for system security.
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4.3 WEB Configuration
DWG2000-1G WEB configuration interface consists of the navigation tree and the detail configuration interfaces.
Figure 4-3-1 WEB introduce
Go through navigation tree, user can check, view modify, and set the device configuration on the right of configuration interface.
4.4 System Information
Figure 4-4-1 system information
System information interface shows the basic information of status information, Mobile information and SIP information.
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4.4.1 System Information
Figure 4-4-2 system information
MAC Address
Table 4-4-1 Description of system information
Displays the current MAC of the gateway, for example: 00-1F-D6-1B-3D-02
Network Mode DWG2000-1G support two types network mode, which is bridge and route modes
Network Shows IP address and subnet mask
DNS Server Displays DNS server IP address in the same network with the gateway
System Up Time shows the time period of the device running. For example,:1h: 20m, 24s
Traffic Statistics Calculates the netflow, including the total bytes of message received and sent.
Version info shows the current firmware version, for example EIA AOS 9.50.34 PCB
4.4.2 Mobile Information
Figure 4-4-3 Mobile information
Display GSM / CDMA channel and network status information, detailed shown as below:
Table 4-4-2 Description of mobile information
Port Numbers of ports of GSM/CDMA.
Type
IMSI
Status
Indicates the current type of network. Such as CDMA or GSM
International Mobile Subscriber Identity, it is the uniquely identifies of SIM card
Indicates the connection status of current GSM / CDMA module
Remaining
Call Duration
Carrier
Limite a call duration to the SIM card, when call duration is out of that duration, the call would be discontinued. This option shows remaining talk time.
Displays the network carrier of current SIM card.
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Signal Quality Displays the signal strength of in each channels of GSM / CDMA.
4.4.3 SIP Information
Figure 4-4-4 SIP information
Displays registration status information with Softswitch platform or SIP Server
Table 4-4-3 Description of SIP information
Port He corresponding number of GSM channel, DWG2000-1G has only 1 port.
SIP User ID SIP registration account of the Softswitch and SIP server provided
SIP User ID Shows the registration status of VoIP channel, including registered and unregistered.
4.5 Network Configuration
The navigation tree of the route mode and bridge mode as below:
Figure 4-5-2 route mode
Figure 4-5-1 bridge mode
In the navigation tree of route mode, will have extra items of “MAC Clone”,”DHCP
Server” ,”DMZ Host” ,”forward Rule”,”Static Route” .
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4.5.1 Local Network
Under the route mode, WAN port connects with ADSL modem, and LAN port connects with local network. It will be used as a small switch when working in brigde mode. In this situation, user just need to configure the WAN parameter and DNS. User also need configure the LAN port if working in route mode. The web interface as bellows:
Figure 4-5-3 WEB interface of bridge mode
Figure 4-5-4 WEB interface of Route mode
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Work Mode
Link Speed & Duplex
Obtain IP Address
Automatically
Use the Following IP
Address
PPPoE
Obtain DNS Server
Address Automatically
Use the Following DNS
Server Addresses
WAN /LAN Port
Table 4-5-1 Description of Local network
Two options of route mode and bridge mode, default is bridge mode
The 5 options are “Auto Detect”,” 10Mbps/Half
Duplex”,”10Mbps/Full Duplex”, “100Mbps/Half Duplex” and
“100Mbps/Full Duplex”. Default is “Auto Detect”
Enable the device obtain IP Address automatically or not. Default is enabling
Configure the "IP Address", "Subnet Mask" and "Default Gateway" by manual
Need ISP offer the account and password. Use this mode when have not router in the local network.
When enable the WAN port option of "Obtain DNS Server Address
Automatically" , which will be enabled subsequently.
Fill in the IP address of "Primary DNS Server" and "Secondary DNS
Server"
WAN port used for connecting the external network , LAN port used for connecting local network
4.5.2 MAC Clone
Figure4-5-5 Configuration of Mac Address Clone
This function can prevent the device against blocking by the carriers. Enable this function on the
“router mode”, the device can be anti-blocked when the carrier to limit the online users by scanning the MAC address.
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4.5.3 DHCP Server
Under “route mode”, DWG works as a router. Config DHCP serive to enable the DHCP service funcition of DWG, then DWG will works as a DHCP server.
Figure 4-5-6 Configuration of DHCP service
IP address Pool
IP Lease Time
Table 4-5-2 Description of DHCP Server
Determines the range of IP addess of other devices in this network
Sets the duration for how long the IP works with the specific IP. If the duration is out of the duration, the IP would be invalid
DNS info will also be allocated to network devices automatically by
DHCP protocol. Generally, there is no need to configure those items.
The subnet mask, gateway
4.5.4 DMZ Host
In some conditions, certain devices in LAN network need to do two-way communication with
WAN network( e.g. certain computer in LAN network need to provide multiple services to WAN network). In this situation, Configure this device as the DMZ host.
Figure 4-5-7 Configuration of DMZ Host
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4.5.5 Forward Rules
In some conditions, certain devices in LAN network need to provide channel communication with
WAN network( e.g. certain computer in LAN network need to provide FTP service of channel 21 to WAN network). In this situation, Configuring forwarding rules to this device is necessary.
The difference between forwarding rules and DMZ host is, DMZ host provides several consecutive channels and communication of all protocols, while forwarding rules provides single or several channels communication based on certain protocol(TCP or UDP). If DMZ host and forwarding rules have conflicts, will be determined by forwarding rules configurations.
Figure 4-5-8 Configuration of forwarding rules
Service Port
IP address
Protocol
4.5.6 Static Route
Table 4-5-3 Description of Forward rules
The Service channel that should be provided to WAN network
IP address is the one of devices in LAN net work
The service protocol(TCP or UDP)
Static route is the route rules in IP communication. Generally speaking, no need to config static route. Configuring static route is necessary in such conditions: when several network segaments exist in LAN network and there’s certain application between these network segament. Please cancel “internet sharing” under “ Network configuration” first, then configure the “static route”.In commly use, please don’t configure static route. If static rule is wrong, the devices may not work.
Figure 4-5-9 Configuration of Static route
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Dest IP Address
Subnet Mask
Nexthop
4.5.7 Qos Parameter
Table 4-5-4 Description of Static Route
Destination IP address the date packets will be sended
The Subnet Mask of Destination IP address
The Next Hop IP address if want to arrive the Destination IP addres
Figure 4-5-10 Qos parameter
If you want to use Qos,please select yes,and click save
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4.6 Mobile Configuration
4.6.1 Basic Configuration
Figure 4-6-1Basic Configuration
Dial Tone Gain
Select Band
Table 4-6-1 Description of Basic Configuration
It is the dial tone volume of call waiting, dial tone of mobile module when call out. Usually adopt the default configuration.
Acording to carrier’s band standards. Standards are as belows:
GSM: 850/900/1800/1900 MHz; CDMA: 800 MHz
Remote API Enable API is provided for third party development with DLL and IAD components. The API includes SMS sending and receiving, USSD sending and receiving. The default is "No"
API Server Address It is the remote IP address who uses API. This is an option when selecting
"Yes" under 'remote API enable"
API Server Port It is the remote channel No. who uses API. This is an option when selecting "Yes" under "remote API enable"
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4.6.2Mobile
DWG2000-1G User Manual
Figure 4-6-2 Mobile Configuration
Table 4-6-2 Description of Mobile Configuration
Mobile Number SIM card user ID of the channel. That must be configured when “one access
No.” funciton enable.
Enable Call
Duration
Limitation
This function is to limit the max call duration of channel. The max call duration is between 1 to 65535 minutes.
Maximum Call
Duration
Defines a value by users. That will limit the SIM/UIM card’s total call duration. After the call duration excesses this value, no call will be initiative by this channel. The value range is 1-65535. If user doesn’t configure this
Minimum
Charging Time
Mobile Number
(Receiving
Alarm) value, Default is no max call duration limits for this channel.
A minimum charging time (in seconds) is defined during which no charging is done at carrier side. If the conversation time is even shorter, the total call duration will not decrease.
The mobile phone No. which used to receive the alarm SMS. Users can get
SMS report of SIM/UIM card status(SIM Remain Time) in DWG.
Alarm Threshold When the SIM remain time is or less than this value, DWG will send the alarm
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(via SMS) SMS to remind the users of the SIM remain time.
Port Description It is the identification mark of SIM/UIM card in the designated SMS report. for Alarm The mobile phone No. of the SIM/UIM card is recommended to use as the port description for alarm, or any other string.
SIM Remain
Time
Restore time
Indicates the current sim remain time. It can’t modified
CLIR
Recovers the SIM remain time to initial value, the Maximum Call Duration.
Caller ID display restrict. This function is used to restrict the mobile phone
No. By adding “#31” before the mobile phone ID, this funciton should be supported by carrier.
Mobile Tx Gain Transits gain of the mobile module, from IP side to PSTN side.
Mobile Rx Gain Receives gain of the mobile module,from PSTN side to IP side.
Detect Reverse This option for CMDA Reverse Polarity detection. Most CDMA operators
Polarity don't offer polarity reverse . So VoIP to mobile, DWG2008 will connect soon.
It doesn't wait mobile side answer.
4.6.3 SIM/UIM Card Lock
Figure 4-6-3 Configuration of SIM/UIM Card Lock
Select Port
SIM Card Lock
PIN Code
Table 4-6-3 Description of Configuration of SIM/UIM Card Lock
Select the Channel No. which need to be locked.
SIM card lock or unlock. Default is “No”.
Correct PIN code is needed to lock or unlock the SIM card.
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4.6.4 PIN Management
DWG2000-1G User Manual
Figure 4-6-4 PIN Management
Detailed description as below:
Table 4-6-4 Description of PIN Management
PIN PIN is the password of SIM card personal identification. In the status of SIM card locked, PIN can be modified to prevent SIM card from being stolen.
Select Port
Old PIN code
New PIN code
Selects the GSM/CDMA channel No.
The previous PIN code
Inputs a new PIN code
4.6.5 SMSC
Figure 4-6-5 SMSC
SMS center of mobile, theoretically, the celluar modular will automatically detect the SMSC number. This configurable option is used in a situation that the SMSC number could not detected by celluar modular. When such case happens, please contact with mobile service provider to identify the SMSC number and then add SMSC number in SMSC configurable web interface.
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4.6.6 SMS
DWG2000-1G User Manual
Figure 4-6-6 SMS sending
Configurations are as below:
Table 4-6-5 Description of SMS sending
Select Port Users can select a defined channel or random channel to send SMS. Input the recevier’s mobile phone No to send SMS.
Addressee Mobile phone No. of the receiver
Message Content of the SMS. The length is limited to 300 characters.
4.6.7 USSD
USSD (Unstructured Supplementary Service Data) is a Global System for Mobile(GSM) communication technology that is used to send text between a mobile phone and an application program in the network. Applications may include prepaid roaming or mobile chatting. USSD is similar with Short Messaging Service (SMS), but unlike SMS. USSD transactions occur during the session only. With SMS, messages can be sent to a mobile phone and stored for several days if the phone is not activated or within range.
Figure 4-6-7 USSD
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Port
Display
Input
4.6.8 Carrier
Table 4-6-6 Description of USSD
Select the GSM channel to send USSD
Display the result of sending USSD
The area to input USSD code
Figure 4-6-8select Carrier
This function is used to select carrier.
Table 4-6-6 Description of select Carrier
Select Port Select GSM channel,default Port 0
Select Mode
Carrier List
There are two mode to select carrier,automatic and manual.
If you select manual mode,you can select carrier from carrier list.
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4.7 Routing Configuration
4.7.1 Routing Parameter
Figure 4-7-1 Routing Parameter
Table 4-7-1Description of Routing Parameter
Tel->IP Parameter Globle parameters, it will take effect while number manipulation configured
Route calls after manipulation
Route calls before manipulation
The parameters indicate that the gateway will select Tel->IP routes after number manipulation completed
The parameters indicate that the gateway will select Tel->IP routes before number manipulation completed
4.7.2 Tel->IP Routing
Figure 4-7-2 Tel to IP Routing
Tel -> IP Routing
Index
Table 4-7-2 Description of Tel to IP Routing
This item uses to configure incoming call routes which can be used for recieve the calls from the GSM.
It uniquely identifies a route. Its value is assigned globally, ranging from 0 to 31.
Description
Source Port
It describes the route for the ease of identification. Its value is character string
It specifies the Port or Port Group which will receive the calls from PLMN
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Source Prefix
Destination Prefix
Destination
All the caller number must match the source prefix. It specifies the source prefix allow to send call out
Any: include anonymous, 0xxxx, 1[2-9]xxxx etc.
0xxxx: consist of some digits such as 015,08,09
1[3-8]6:consist of some prefix, include 136,146,156,166,176, 186
All the called number must match the destination prefix, the call prefix indicates the connected number
Any: include anonymous, 0xxxx, 1[2-9]xxxx etc.
0xxxx: consist of some digits such as 015,08,09
1[3-8]6:consist of some prefix, include 136,146,156,166,176, 186
Its specifies destination Port or Port Group
Figure 4-7-3 Tel to IP routing Modify
It's a default route configured in gateway. It allows any number from source port 0 send call to SIP server with any prefix.
Figure 4-7-3 Tel to IP routing Modify
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Add a GSM to VoIP route. It indicates that the calls coming from Port Group 31<Unicom> will match the prefix ”x.”, ”x.” is a wildcard string which will match any prefix except ”anonymous” calls. Meanwhile sending the calls destination IP 13<eia> if called number match with destination prefix ”00”.
Figure 4-7-3 Tel to IP routing Modify
Add GSM to GSM route,its mainly used for saving the cost between two carriers.It indecates that calls coming from Port 0 will match the prefix 13[58], ”13[58]” include prefix 135 and 138, caller number can’t match prefix 135 and 138 will reject by gateway. Meanwhile sending the calls to
Port Group 31<Unicom> if called number match with prefix 133.
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4.8 Manipulaton Configuration
4.8.1 IP->Tel Destination Numbers
Figure 4-8-1 IP->Tel destination numbers manipulation
Table 4-8-1 Description of IP->Tel destination numbers manipulation
IP->Tel numbers destination manipulation
Index
Description
Source IP
Source Prefix
Destination Prefix
It is an optional configuration item, and is used to add IP->Tel number change data.
The IP->Tel Manipulation defined the rules of add, and deletion of called numbers, which are referenced by IP->Tel routing.
It uniquely identifies a route. Its value is assigned globally, ranging from 0 to 31.
It describes the route for the ease of identification. Its value is character string
It specifies the source IP which will send the calls to gateway
Any: any IP address
IP: specific an IP address
IP Group: specific an IP group
All the caller number must match the source prefix. It specifies the source prefix allow to send call out
Any: include anonymous, 0xxxx, 1[2-9]xxxx etc.
0xxxx: consist of some digits such as 015,08,09
1[3-8]6:consist of some prefix, include 136,146,156,166,176, 186
All the called number must match the destination prefix, the call prefix indicates the connected number
Any: include anonymous, 0xxxx, 1[2-9]xxxx etc.
0xxxx: consist of some digits such as 015,08,09
1[3-8]6:consist of some prefix, include 136,146,156,166,176, 186
Its specifies destination Port or Port Group Destination
Stripped Digits from
Left
Stripped Digits from
Right
Prefix to Add
Suffix to Add
It specifies the length of the digits to be deleted from left
It specifies the length of the digits to be deleted from right
Add the new digits in front of the original number
Add the new digits at the end of the original number
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Add an IP->Tel Manipulation, to change the called number from 2547888888 to 07888888
Figure 4-8-2 IP->Tel destination numbers manipulation modify
It indecates that calls coming from IP Group will match the prefix ”any”, and the called nubmer whom match with the prefix ”2547” will delete 3 digits in front of it and replace it by digit ”0”.
4.8.2 Tel->IP Source Numbers
Figure 4-8-3 Tel->IP destination numbers manipulation
Table 4-8-2 Description of Tel->IP destination numbers manipulation
Tel->IP destination numbers manipulation
It is an optional configuration item, and is used to add IP->Tel number change data.
The IP->Tel Manipulation defined the rules of add, and deletion of called numbers, which are referenced by IP->Tel routing.
Index It uniquely identifies a route. Its value is assigned globally, ranging from 0 to 31.
Description It describes the route for the ease of identification. Its value is character string
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Source Prefix
All the caller number must match the source prefix. It specifies the source prefix allow to send call out
Any: include anonymous, 0xxxx, 1[2-9]xxxx etc.
0xxxx: consist of some digits such as 015,08,09
1[3-8]6:consist of some prefix, include 136,146,156,166,176, 186
Destination Prefix
All the called number must match the destination prefix, the call prefix indicates the connected number
Any: include anonymous, 0xxxx, 1[2-9]xxxx etc.
0xxxx: consist of some digits such as 015,08,09
1[3-8]6:consist of some prefix, include 136,146,156,166,176, 186
Its specifies destination Port or Port Group Destination
Stripped Digits from
Left
It specifies the length of the digits to be deleted from left
Stripped Digits from
Right
Prefix to Add
Suffix to Add
It specifies the length of the digits to be deleted from right
Add the new digits in front of the original number
Add the new digits at the end of the original number
Number of Digits to
Leave from Right
It specifies the number of Digits to Leave from Right
Example
Add an IP->Tel Manipulation, to change the called number from 2547888888 to 07888888
Figure 4-8-4 Tel ->IP destination numbers manipulation add
It indecates that calls coming from IP Group will match the prefix ”any”, and the called nubmer whom match with the prefix ”2547” will delete 3 digits in front of it and replace it by digit ”0”.
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4.8.3 Tel->IP Destination Numbers
Figure 4-8-5 Tel->IP destination numbers manipulation
Table 4-8-3 Description of Tel->IP destination numbers manipulation
Tel->IP destination numbers manipulation
It is an optional configuration item, and is used to add IP->Tel number change data.
The IP->Tel Manipulation defined the rules of add, and deletion of called numbers, which are referenced by IP->Tel routing.
Index It uniquely identifies a route. Its value is assigned globally, ranging from 0 to 31.
Description It describes the route for the ease of identification. Its value is character string
Source Prefix
Destination Prefix
Destination
Stripped Digits from
Left
Stripped Digits from
Right
Prefix to Add
Suffix to Add
Number of Digits to
Leave from Right
All the caller number must match the source prefix. It specifies the source prefix allow to send call out
Any: include anonymous, 0xxxx, 1[2-9]xxxx etc.
0xxxx: consist of some digits such as 015,08,09
1[3-8]6:consist of some prefix, include 136,146,156,166,176, 186
All the called number must match the destination prefix, the call prefix indicates the connected number
Any: include anonymous, 0xxxx, 1[2-9]xxxx etc.
0xxxx: consist of some digits such as 015,08,09
1[3-8]6:consist of some prefix, include 136,146,156,166,176, 186
Its specifies destination Port or Port Group
It specifies the length of the digits to be deleted from left
It specifies the length of the digits to be deleted from right
Add the new digits in front of the original number
Add the new digits at the end of the original number
It specifies the number of Digits to Leave from Right
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Example
Add an IP->Tel Manipulation, to change the called number from 2547888888 to 07888888
Figure 4-8-6 Tel->IP destination numbers manipulation
It indecates that calls coming from IP Group will match the prefix ”any”, and the called nubmer whom match with the prefix ”2547” will delete 3 digits in front of it and replace it by digit ”0”.
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4.9 Operation
4.9.1 IP->Tel Operation
Figure 4-9-1 IP->Tel Operation
IP->Tel Operation
Table 4-9-1 Description of IP->Tel Operation
It is an optional configuration item. Operation configuration essentially involves allow, barring some IP and IP Group send calls to certain numbers. It includes: forbid call, call allowance, auto call, and password authentication.
It uniquely identifies a route. Its value is assigned globally, ranging from 0 to 31. Index
Source IP
Source Prefix
Destination Prefix
Operation
It specifies the source IP which will send the calls to gateway
Any: any IP address
IP: specific an IP address
IP Group: specific an IP group
All the caller number must match the source prefix. It specifies the source prefix allow to send call out
Any: include anonymous, 0xxxx, 1[2-9]xxxx etc.
0xxxx: consist of some digits such as 015,08,09
1[3-8]6:consist of some prefix, include 136,146,156,166,176, 186
All the called number must match the destination prefix, the call prefix indicates the connected number
Any: include anonymous, 0xxxx, 1[2-9]xxxx etc.
0xxxx: consist of some digits such as 015,08,09
1[3-8]6:consist of some prefix, include 136,146,156,166,176, 186
Its specifies number analysis rule
Forbid call
Allow call
Auto call
Password authenticate
It describes the route for the ease of identification. Its value is character string Description
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Example
Index 31: barring the certain calling number from IP 14<elastix>
Figure 4-9-2 IP->Tel Operation Modify
It indicates that calling party from IP 14<elastix> matched prefix 2877, and also called party matched prefix 07 are not allowed call out. The calls match this rule will be rejected by gateway.
Index 29: definite a rule for IP 17<FreeSentral> that all the calls must go with valid password authentication.
Figure 4-9-3 IP->Tel Operation Modify
4.9.2 Tel->IP Operation
Figure 4-9-4 Tel->IP Operation
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Description
Tel->IP Operation
Table 4-9-2 Description of Tel->IP Operation
It is an optional configuration item. Operation configuration essentially involves allow, barring some IP and IP Group send calls to certain numbers. It includes: forbid call, call allowance, auto call, and password authentication.
Index It uniquely identifies a route. Its value is assigned globally, ranging from 0 to 31.
Source IP
Source Prefix
Destination Prefix
Operation
It specifies the source IP which will send the calls to gateway
Any: any IP address
IP: specific an IP address
IP Group: specific an IP group
All the caller number must match the source prefix. It specifies the source prefix allow to send call out
Any: include anonymous, 0xxxx, 1[2-9]xxxx etc.
0xxxx: consist of some digits such as 015,08,09
1[3-8]6:consist of some prefix, include 136,146,156,166,176, 186
All the called number must match the destination prefix, the call prefix indicates the connected number
Any: include anonymous, 0xxxx, 1[2-9]xxxx etc.
0xxxx: consist of some digits such as 015,08,09
1[3-8]6:consist of some prefix, include 136,146,156,166,176, 186
Its specifies number analysis rule
Forbid call
Allow call
Auto call
Password authenticate
It describes the route for the ease of identification. Its value is character string
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4.10 IP Trunk Configuration
4.10.1 IP Trunk
Figure 4-10-1 IP Trunk
IP Trunk
Table 4-10-1 Description of IP Trunk
Add remote IP of softswitch, SIP server which will send call traffics to gateway.
It uniquely identifies a route. Its value is assigned globally, ranging from 0 to 31. Index
Description It describes the route for the ease of identification. Its value is character string
IP
Port
It is an interworking parameter between the remote Softswitch and the SIP server.
It specifies the IP address of the peer equipment.
It is an interworking parameter between the remote Softswitch and the SIP server.
It specifies the SIP port number of the peer equipment
Example
To add a remote IP of Softswitch, set “index” to “31”, SIP port number “5060”
Figure 4-10-2 IP Trunk Modify
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4.10.2 IP Trunk Group
Figure 4-10-3 IP Trunk Group
IP Trunk Group
Index
Description
IP
Table 4-10-2 Description of IP Trunk Group
This configuration is optional, and is used to add the IP that have the same attributes to an IP group. The IP group will referenced by IP->Tel routing and number manipulation.
It uniquely identifies a route. Its value is assigned globally, ranging from 0 to 31.
It describes the route for the ease of identification. Its value is character string
It specifies the IP will add to IP group
Example
To add an IP group, set IP “10, 14, 17” to IP group 18
Figure 4-10-4 IP Trunk group modify
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4.11 System Configuration
4.11.1 System Configuration
Figure 4-11-1 System Configuration
Voice Prompt Language
Table 4-11-1 Description of System Configuration
Configure the voice prompt of DWG., e.g. configure voice prompt of IP address success or failure. DWG supports English and Chinese. Users can customize other languages. The default setting is in Chinese.
Provision configuration Provision is used to maintain the devices. E.g. Provision can config, update and remote manage the devices in bulk.
Primary Provision Server IP This is provided by carrier. Keep the default value if carrier don’t provide this value.
Primary Provision Server This is provided by carrier. Default is 80.
Port
Secondary Provision Server This is provided by carrier. Keep the default value if carrier don’t
IP provide this value.
Secondary Provision Server This is provided by carrier. Default is 80.
Port
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Provision Check Interval
Enable NTP
Primary NTP Server IP
Secondary Provision NTP
Server IP
Time Zone
Default is 4 hours.
NTP enable switch
Can keep the default
Can keep the default
4.11.2 Service Configuration
The default is GMT +8:00, the user can adjusted accordingly according to their area
Service Configuration is used for configuring voice calls and some small businesses, such as Call
Progress Tone, codec, silence suppression, * service, the second dial and so on
Figure 4-11-2 Service Configuration
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LOCAL RTP PORT
Channel
Enable Silence
Suppression
Call Progress Tone
Table 4-11-2 Description of Service Configuration
Means the initial allocation of Channel when RTP voice stream transmit in the IP network , in general, using the factory default values. When there are multiple DINSTAR series voice products, and the network gateway or router’s NAT with loopholes, user can try changing this item
Enable the "silence suppression" almost no impact on call quality, and can save about half of the bandwidth.
Each country has its different call progress tone required standards, such as busy tone, ring back tones and ring tone standards, users can select the area standard from here .
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Preferred Coders
Enable
Incoming
Configuration
Means the code format when Voice transfer on IP network, support
PCMA, PCMU, G.723.1 andG.729AB.
Note: when the preferred codec switch between G.723.1 and G.729AB ,
System will automatically reset
PSTN
Means when call from PSTN side, you can dial the function keys for check number, setting IP and so on function
Enable Auto Outgoing
Means when call out , whether by ordinal or polling pick to Select a
Channel, this feature are generally used for when use the same SIP User
Routing
ID to register or use as trunking mode
This function will be displayed only when select "Enable Auto Outgoing
IP to PSTN One Stage
Routing" function, the User ID will be sent directly to PSTN, for
Dialing example: the user calls 6715, the device will sent 6715 User ID to PSTN
Setting is yes, when through the PSTN calls to the Channel, the device
Play Voice Prompt for will with the clew tone, the default is "Please dial the extension User
PSTN Incoming Calls
ID"; setting to No, the device will with dial tone
For Example, the phone A from PSTN side call
DWG2000-1G/DWG2004/DWG2008 Channel SIM card corresponding
User ID, the Channel's SIP User ID is C, Channel hook and then call B,
Send Original Caller
ID for PSTN Incoming when "Send Original Caller ID for PSTN Incoming" setting is Yes, the
Caller User ID that send to B will be A, when "Send Original Caller ID for
PSTN Incoming" setting is No, the caller User ID that sent to B will be
C(except for anonymous outgoing )
DTMF
DWG2000-1G/DWG2004/DWG2008 support RFC2833 and SIGNAL two ways. DTMF INTERVAL range is 50 ~ 800ms, DTMF VOLUME can use the default Configuration
(Simple Traversal of UDP over NATs,NAT’s UDP simple cross) is a
Enable STUN network protocol. It is allowed to stay behind the NAT (or multiple NAT) client part to identify their clients’ public address, found himself after what Type of NAT and NAT for a particular Channel is bound to a local
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Internet terminal Channel. This information is used for two host to set up
UDP communication behind the same NAT router. The agreement defined by the RFC 3489
Select the name under the special needs , the most common way is use the
CLID Mode default User ID
Allow Outgoing Calls
Refer to "SIP Configuration" -> "Is register" . If "Is register" setting is no, without Registration this option need set Yes ,to avoid that the devices can not call out
Allow Incoming Calls Refer to "SIP Configuration" -> "Is register" . If "Is register" setting is no, without Registration this option need set Yes ,to avoid that the devices can not call in
Allow Anonymous
Outgoing Calls
The incoming anonymous calls will be rejected
Use # as End Key
In General, SIP phones are based on # as the end, if this option is set to
No, the dial-up will end expires dial-up time
Bit of between the dialing time ,over the time will be seem as end of dial
Inter digit Timeout
4.11.3 SIP Configuration
Figure 4-11-3 SIP Configuration
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SIP
Configuration
Used for Configuring VoIP channel, add SIP Registry Platform and local SIP
Channel, and configure SIP protocol and other related information
SIP Server
Address
Used for configure SIP server address and Channel, the address can be IP
Address, also can be a domain name (DNS should to be able to resolution), the details please advisory the service provider
SIP Proxy Port Port default setting is 5060. For details, please consult the service provider
Outbound proxy, it mainly used in firewall / NAT environment. That make the
Outbound Proxy
Address signaling and media streams are able to penetrate the firewall, the details please advisory the service provider
Outbound Proxy
Port
Outbound proxy port number, the details please advisory the service provider
Use Random
Port
Set the local monitor SIP port(fixed or random), random is every time you start the device will random Select a free SIP port Monitor
Is Register
Default set yes, if you want the device can make a call without register, set
No, Also enable the "Allow Outgoing Calls without Registration" and "Allow
Incoming Calls without Registration" function
Register Interval Means how often the equipment will register once to the SIP server/proxy
T1
T2
T3
Used to define the SIP protocol T1 timer value, default is 500ms
Used to defines the SIP protocol timer values, default value is 4000ms
Keep alive
Interval
Used to define the T2 timer value in SIP protocol, the default is 5000ms
Used to keep communicate between equipment and the SIP server that make the device is in the best available Registered. In general, using the factory default values
4.11.4 Port Configuration
Port Configuration is used to configure ports’ gain, Off-hook Auto-Dial, etc.
Figure 4-11-4 Port Configuration
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Port Configuration
ALL ports register used same user ID
SIP User ID
Table 4-11-3Description of Port Configuration
Used to configure ports’ gain, Off-hook Auto-Dial, etc.
The default is not, each SIP account, if set yes ,all the port will use user
ID of port0, when the call ring in sequence
Is the account used for registration, equipment port's unique identifier,
"Authenticate ID" is equivalent to show the name, "Password" is register
Password, which no passward can no fill, the details please contact the
Tx Gain
Rx Gain
Offhook Auto-Dial service provider
Refers to the call volume that from himself during a call to the end users, adjust the "Tx Gain" will affect the voice volume of the end user, the default value is 0
Refer to the call volume from the remote end user to ourself volume, adjust the "gain acceptance" will affect the voice volume we will heard, the default value is 0.
Hotline service.when PSTN part client calls to this port,will auto forward to the hotline User ID. If no need this feature, just left it blank
Auto-Dial Delay Time
Offhook Auto-Dial delay time, the range is 0-10 seconds
4.11.5 Digit Map
Figure 4-11-5 Digit map
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Digit Map Syntax:
1. Supported objects
Digit: A digit from "0" to "9".
Timer: The symbol "T" matching a timer expiry.
DTMF: A digit, a timer, or one of the symbols "A", "B", "C", "D", "#", or "*".
2. Range []
One or more DTMF symbols enclosed between square brackets ("[" and "]"), but
only one can be selected.
3. Range ()
One or more expressions enclosed between round brackets ("(" and ")"), but
only one can be selected.
4. Separator
|: Separated expressions or DTMF symbols.
5. Subrange
-: Two digits separated by hyphen ("-") which matches any digit between and
including the two. The subrange construct can only be used inside a range
construct, i.e., between "[" and "]".
6. Wildcard
x: matches any digit ("0" to "9").
7. Modifiers
.: Match 0 or more times.
8. Modifiers
+: Match 1 or more times.
9. Modifiers
?: Match 0 or 1 times.
Example:
Assume we have the following digit maps:
1. xxxxxxx | x11
and a current dial string of "41". Given the input "1" the current dial
string becomes "411". We have a partial match with "xxxxxxx", but a
complete match with "x11", and hence we send "411" to the Call Agent.
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2. [2-8] xxxxxx | 13xxxxxxxxx
Means that first is "2","3","4","5","6","7" or "8", followed by 6 digits;
or first is 13, followed by 9 digits.
3. (13 | 15 | 18)xxxxxxxxx
Means that first is "13","15" or "18", followed by 8 digits.
4. [1-357-9]xx
Means that first is "1","2","3" or "5" or "7","8","9", followed by 2 digits.
4.12 Tools
4.12.1 Firmware Upload
Equipment upgrades can through Dinstar’s softswitch platform, when the device disconnects with
Dinstar’s softswitch platform or some special circumstances. The firmware is able to upload locally.
Figure 4-12-1 Firmware upload
Select the upgrade program under correct directory services, and then click upload will complete upgrade the firmware.
NOTE: during the upgrade process, please do not swtich off the power supply, equipment may paralyze.
4.12.2 IVR Voice Prompt Upload
By default, when PSTN call incoming, the system will play the default IVR, and also the user can load custom IVR.
Figure 4-12-2 IVR Voice Prompt Upload
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NOTE: the customize voice files can be recorded using Windows recording programs, the sound format is 8000Hz, 16 bit sampling in mono, with WAV format, size of files can not exceed 190KB
4.12.3 Data Backup
Figure 4-12-3 Data backup
Click 'Backup' for download configuration file to your computer.
4.12.4 Data Restore
Figure 4-12-4 Data restore
Send data file from your computer to the device
4.12.5 Syslog Parameter
Figure 4-12-5 Syslog parameter
Table 4-12-1 Description of Syslog Parameter
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Enable SysLog select yes to enable syslog client function
Server Address Fill in the Syslog server IP Address here
Syslog Level There are five level of syslog:NONE、DEBUG、NOTICE、WARNING、
ERROR, we urge you to select DEBUG.
Send CDR If you select yes, DWG will send CDR to syslog server.
4.12.6 Login Password
Figure 4-12-6 IVR Voice Prompt Upload
When using web or telnet Configuration, please enter default user name and password. User can modify the login name and password.
4.12.7 Factory Reset
Figure 4-12-7 Factory Reset
Be careful do this operation, after restore factory setting, all the parameters will be changed to the factory default.
4.12.8 Restart
Figure 4-12-8 Restart
When system restarts, user click RESET button on the web.
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DWG2000-1G User Manual
5. FAQ
5.1 Device have been connected to network physically, but the network cannot be connected or network communication is not normal
1) Make sure the network cable is ok or not , can through view the device WAN port or LAN port indicator light to determine the physical connection is working or not;
2) Make sure the connected network devices (router, switch or hub) support 10M/100M adaptive, if not, connect the Equipment directly to PC, landing WEB and in the "local connection"
Configuration interface Select the correct Ethernet Work Mode;
3) Check the Network Configuration, if the Configuration is incorrect, please re-Configuration. If you are using DHCP mode, check DHCP Server is working properly;
4) Check whether there is a LAN device conflict with the exists IP ADDRESS.
5.2 Equipment can not register
If the Run LED does not flash mean unregistered
1) Check the network connection is working (see above section), whether the Configuration is correct;
2) Check whether the LAN firewall setting is inappropriate (such whether limit the network communication); If it is, there are two ways to try to resolve;
3) Ask network administrators to open limitation with the equipment’s network communications
(it is a special equipment, not afraid of virus attacks); (2) try to enable the equipment tunnel
(through the WEB for Configuration, Also, please NOTE, open the tunnel will impact voice quality, Please do not enable the tunnel as far as possible, reference WEB Configuration Interface
Description section);
4) Check whether the Local Network to the SIP PROXY platform network environment is relatively poor or not, and if so, please check Local Network or contact the service provider;
5) if go through those steps, the device still be in trouble, please contact the equipment provider;
5.3 When calling out, the callee’s phone shows wrong caller ID:
1) Ask the callee checks whether the device is failure or device battery power is low
2) Make sure the callee has been subscribed called User ID display service
3) If only part of the caller User ID with this problem, please contact the telecom carrier.
5.4 when calling in, the caller always hears a busy tone
Make sure Enable DND(Do-not-Disturb) in system
5.5 sudden interruption during a call
1) make sure whether is human error caused the problem
2) Make sure with the account balance or lack of disruption caused the call disconnected
3) Make sure whether there is interference with the fax tone or equipment busy tone, these interference may lead to calls dropped
4) Make sure whether the LAN equipment such as gateway or router fails, user can try to restart the gateway or router
_______________________________________________________________________________
Dinstar Technologies Co., Ltd.
51
VoIPon www.voipon.co.uk [email protected] Tel: +44 (0)1245 808195 Fax: +44 (0)1245 808299
DWG2000-1G User Manual
5.6 voice single-pass, double-barrier or poor quality
1) Make sure the equipment is working properly with grounded power
2) Check the device network connection is in working status
3) Ask network administrators to open limitation with the equipment’s network communications
(it is a special equipment, not afraid of virus attacks); (2) try to enable the equipment tunnel
(through the WEB for Configuration, Also, please NOTE, open the tunnel will impact voice quality, Please do not enable the tunnel as far as possible, refer WEB Configuration Interface
Description section)
4) Make sure the LAN equipment is working, user can try to restart the gateway or router to solve the problem
5) Check whether there is more than one DINSTAR series products in LAN network: some gateways or routers, processing network packet is vulnerable (for example, to multiple network devices or the same protocol network communication, NAT allocated the same conversion communications Channel). If there is such a case, suggest replacing a router or specify each voice gateway with different LOCAL RTP PORT Channel (refer to the base WEB Configuration interface section)
6) Check the equipment network environment for the softswitch platform, monitor the network condiation, make sure the network is solid
6. Glossary
GSM: Global System for Mobile Communications
CDMA: Code Division Multiple Access
FMC: Fixed Mobile Convergence
SIP: Session Initiation Protocol
MGCP: Media Gateway Control Protocol
DTMF: Dual Tone Multi Frequency
USSD: Unstructured Supplementary Service Data
PSTN:Public Switched Telephone Network
STUN: Simple Traversal of UDP over NAT
IVR: Interactive Voice Response
IMSI: International Mobile Subscriber Identification Number
IMEI: International Mobile Equipment Identity
DMZ: Demilitarized Zone
_______________________________________________________________________________
Dinstar Technologies Co., Ltd.
52
VoIPon www.voipon.co.uk [email protected] Tel: +44 (0)1245 808195 Fax: +44 (0)1245 808299
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