DVG-4032S VOIP Gateway User Manual Version 1.
Contents 1. Introduction....................................................................................................1 Product Overview .......................................................................................................................................... 1 Product Features........................................................................................................................................... 2 Hardware Description..................................................
2 Software Upgrade ....................................................................................................................................... 44 Logout ......................................................................................................................................................... 44 5. Coding Principle ..........................................................................................46 Instruction................................................................
1 1. Introduction Product Overview The stand-alone VoIP Gateway carries both voice and facsimile over the IP network. It supports SIP industry standard call control protocol to be compatible with free registration services or VoIP service providers’ systems. It works in two different modes: UA (User Agent) or Server. As a standard user agent, it is compatible to all well-known Soft Switches and SIP proxy servers.
2 Product Features ● SIP (RFC 3261) compliant ● Optional server enables small businesses to build up private VoIP network (SIP model) ● QoS support guarantees voice bandwidth in a busy network ● Supports IP TOS (Type Of Service) ● T.30 (G III) / Real Time T.38 / Secured T.
3 Hardware Description Front Panel Power Indicator: Green light indicates a normal power supply. Run Indicator: Blinking green light indicates normal operation. Alarm Indicator: When the system starts up, the red light will blink. It also indicates the gateway’s abnormal operation. Phone 1 – Phone 16 stands for Port 1 – Port 16: Connect to your analog telephone, Line 1 – Line 16 stands for Port 17 – Port 32: Connected to your original telephone line on the wall jack with RJ-11 cable.
4 2. Installation and Applications Network Interface The network interface is divided into 3 basic modes as described below: Gateway can be assigned with a Public IP Address Gateway can be built under the existing NAT Gateway Assigned with a Public IP Address The gateway will have a Public IP address for Internet connection regardless of whether it is a static IP address, DHCP (using a Cable Modem), or PPPoE (Dialup / ADSL).
5 Gateway in a NAT network The gateway uses a virtual IP address and the IP sharing function of other systems to connect to the Internet. LAN IP address of IP sharing Please avoid IP address 192.168.0.1-192.168.8.254 (You may need to change the settings of IP sharing or change SIP series Gateway LAN Port IP address) Gateway IP Settings Set as static IP address, and assign the LAN IP address of the IP sharing to the Default Gateway.
6 Telephone Interface Description Example for Phone Ports: DVG-4032S connecting directly to phone sets After connecting telephone sets to P1-P8, users can make direct calls, (P1-P8 are FXS interfaces). Each set acts as an independent extension line. Integrating the DVG-4032S with PBX P1-P16 is FXS interfaces, and some of them can be connected to telephone sets for direct calls. Others can be connected to the PBX so other extension lines can make VoIP calls.
7 3. Setting the Gateway through IVR VoIP transmits voice data (packet) via the Internet to achieve telecommunications. This means that the telecommunication quality is closely related to the whole network environment. If any one of the telecommunicating parties has insufficient bandwidth or frequent packet loss, the telecommunication quality will be poor. Therefore, an excellent telecommunication can only be created when Gateway is connected to the Internet and when network environment is stable.
8 Instructions FXS Port: Connected to telephones. To enter IVR mode, enter “ * * password #” after hearing the dial tone. When you hear a second dial tone, the system is in IVR mode, enter the function code. (Please refer to the Advanced Settings on page 35 for these codes) FXO Port: to use IVR functions, dial the phone number of FXO Port using an external line. You will hear the instruction “enter value”, and then enter a PIN number. The factory default code is blank. Enter “**#” as above.
9 IVR Functions Table: Function Code Description Example 111/101 WAN Port IP address Set/Query 112/102 WAN Port Subnet Mask Set/Query 113/103 WAN Port Default Gateway Set/Query Use in conjunction with function code 114, select 1 for a Static IP function. 114/104 Current Network IP Access Set/Query (1: Static IP, 2.DHCP, 3.
10 Function Code Description 215/205 Set/Query Gateway Telephone Number (Representative Number) 216/206 Set/Query the extension number of Line 1. Example 217/207 109 Restoring factory default IP address configuration 409 Restoring factory default settings 509 Save settings 900 Setting IVR and the language used on the Web GUI (1: English, 2: Traditional Chinese, 3: Simplified Chinese) 209 Soft Upgrade A static IP address for WAN Port IP：192.168.1.2 Mask：255.255.255.0 Gateway：192.168.1.
11 IP Configuration Settings—Setting IP Configuration of WAN Port Static IP Settings NOTE: Complete static IP settings should include a static IP (Option 1 under114), IP address (111), Subnet Mask (112), and Default Gateway (113). Please contact your local Internet Service Provider (ISP) if you have any questions. Function Command Select a Static IP After entering IVR mode, dial 114. After hearing “Enter value”, dial 1 (to select static IP) IP address Settings After entering IVR mode, dial 111.
12 PPPoE Account Settings After entering IVR mode, dial 121. After hearing “Enter value”, enter the account number, followed by ”#”. Example: If the account is “84943122 @ hinet.net”, please enter 08 04 09 04 03 01 02 02 71 48 49 54 45 60 72 54 45 60 #. Please note that it is necessary to enter two digits for each character/number; for example, enter “01” for “1” and “11” for “A”. PPPoE Password Setting After entering IVR mode, dial 122 after hearing “enter value” followed by “#”.
13 PPPoE Character Conversion Table Number Input Key Upper Case Input Key Letter Lower Case Input Key Letter Symbol Input Key 0 00 A 11 a 41 @ 71 1 01 B 12 b 42 • 72 2 02 C 13 c 43 ! 73 3 03 D 14 d 44 " 74 4 04 E 15 e 45 $ 75 5 05 F 16 f 46 % 76 6 06 G 17 g 47 & 77 7 07 H 18 h 48 ' 78 8 08 I 19 i 49 ( 79 9 09 J 20 j 50 ) 80 K 21 k 51 + 81 L 22 l 52 , 82 M 23 m 53 - 83 N 24 n 54 / 84 O 25 o 55 :
14 4. Setting a Gateway with WEB Browser The gateway allows users to make settings using a web browser. After opening a browser, enter Gateway’s IP address as the website address in order to enter the Web configuration screen as shown in the following diagram. The factory default WAN IP address for Gateway is 192. 168. 1. 2.You can also enter ”101” from the handset to inquire about the current WAN Port IP address. The factory default LAN Port IP address is 192.168.8.254.
15 Current WAN IP Address: The IP address of WAN port. Listen Port UDP: It is not necessary to change the protocol of the communication port used by DVG-3016S. RTP Starting Port UDP: The initial value of port number for transmitting voice data among Gateway(s). Each line requires 2 ports. It is not necessary to change these. For example, If the starting port is 9000, then Line 1 is using 9000 and 9001, and Line 2 is using 9002 and 9003, and so forth.
16 IP Configuration (Setting WAN Port) There are four methods of obtaining a WAN port IP address: 1. Static IP 2. DHCP, means a Dynamic IP (Cable Modem) 3. PPPoE (Dialup ADSL) 4. PPTP. Using the DHCP and PPPoE for obtaining an IP address may vary. If not familiar with the network connection, please contact your local ISP. Setting Dynamic IP (DHCP) Click “DHCP” to obtain a Dynamic IP address, then click the “Accept” button at the bottom of the screen.
17 PPTP※ Select “PPTP” and enter the IP Address, Subnet mask, PPTP Server, PPTP ID and Password. Then click the “Accept” button at the bottom. Save the settings, and then restart the system. The system will take about 40 seconds to restart. If not familiar with the network connection, please contact your local ISP.Domain Name Server. BigPond (for Australia use only) Click “BigPond Cable” Enter User Name and Password. Login Server is optional. Then click the “Accept” button at the bottom.
18 Clone MAC Some Internet Service Providers (ISP) assigns the bandwidth via the MAC (Media Access Control) Address. You can click the " Clone" button to copy the MAC address of the Ethernet Card installed in the computer used to configure the device. It is only necessary to fill in the field if required by your ISP. The “Your MAC Address” will be blank as you log in through WAN port.
19 Network Settings (LAN) Network Settings (LAN): Gateway LAN Port IP address and the subnet mask value. Please note that Gateway is built under NAT: Gateway LAN Port IP address cannot be in the same section as the NAT LAN Port IP address, or else it is unable to make or receive calls. For example, if the NAT LAN Port IP address is 192.168.8.1, then Gateway LAN Port cannot be ranged between 192.168.8.1 ~ 192.168.8.254. If so, please change the LAN port IP address, (e.g. setting the IP address to 192.168.
20 NAT/DDNS NAT Traversal If a Gateway is set up under an IP sharing you can select NAT or STUN protocol. NAT: The IP address used by Gateway should be a virtual address. Further, users must set the Virtual Server Mapping in the NAT Server (A virtual server is defined as a Service Port，and all requests to this port will be redirected to this specified the server IP address ).
21 These settings are only necessary when Gateway is set up under a NAT, which not only uses a dynamic IP address but also does not support DDNS. Choose a DDNS Server: The current system allows users to choose either DynDNS、TZO、 3322.org、PeanutHull or a private server. Please apply for a user account before choosing a service provider. Server: Set up the IP address or URL (Uniform Resource Locator) of the DDNS Server.
22 Telephony Settings Prefix Number Rules Trunk Dial Out Verify/ Trunk Dial Out Replace: VoIP gateway will check (verify) the dial out prefix from dial out numbers and change (replace) the prefix to transit out through FXO port. For example: If you transit out with 01907123456, the system will trans to 190601 907123456. If you transit out with 008621123456 the system will replace it with 190200 8621123456. Trunk Dial Out Deny: The system will deny the call with the leading number filled in this column.
23 Anonymous Caller ID (CLIR): When enabled, the caller’s phone set will not display your number. Note: If you register the gateway to a Proxy, you may be unable to make a call. This is due to the fact that the gateway doesn’t send the number for authorization. Anonymous Transit in W/O Caller ID: FXO won’t detect caller ID, and the gateway will dial out with anonymous caller identification. If the call needs caller ID to be identified for Proxy, Proxy will reject this call without caller id.
24 Dial-out Prefix: It is the number dialed automatically by the system when the FXO interface diverts a call to the PSTN by VoIP. FXO Line Default Dial-Out: To set the dial-out number when this FXO line is indicated. Example: If PBX extension needs to dial “0” to make a PSTN call, and the FXO are connected to PBX extension. In this case, the Dial-out prefix should be set to “0”.
25 If the user at Phone 1 (Port 1) of this system wants to assigns Line 8 (FXO) to make a call, he/she can dial 708 22520199. If this item isn’t checked, the gateway will select a line automatically to call out from. For example, dial 22520199 without adding the extension number of the FXO port. Pick up Line by Dialing Extension Number: Allows user to dial just the FXO extension – 708 - to use when the PSTN line is connected on the FXO port. If you are registered to a Proxy, it MUST be checked.
26 Internet. The default is disabled. Session Refresh Request: to send the packet of UPDATE or re-INVITE to Session Refresher: It is the gateway’s role in Session Timer. UAS is an originator, and UAC is a replier. Enable P-Assert: It is for caller id protect. Privacy Type: Privacy requested for Third-Party Asserted. SIP Message Resend Timer Base: SIP packet will resend if response didn't arrive in the base time set in this column.
27 Area Code: Please enter the area code. E.164 Numbering: To invite Proxy to follow the E.164 rule. It depends on the Proxy. If you fail to make a call, please contact your ITSP. Enable Support of SIP Proxy Server / Soft Switch: Enable the functions to inter-work with Proxy Server / Soft Switch. When SIP Proxy 1 and 2 are enabled, the system will register to SIP Proxy 2 after all lines are failed to register to SIP Proxy 1. SIP Proxy 2 is a backup system.
28 contact your ITSP. Bind Proxy Interval for NAT: This function is able to keep the binding is existed when the gateway is behind NAT and SIP Proxy is not able to keep the binding. Initial Unregister: After rebooting, it is unregistered first and then do the general registry process. Enable Message Waiting Indication: The system will play a tone to remind users that there are messages in SIP Server. Proxy-Require: Some SIP Sever need SIP UA to add this header to it's sip message.
29 with them, and according to RFC standards. If any registration problem occurs, please consult your Proxy Server provider. NOTE: When you register with a Proxy Server, dialing principles may vary with different Proxy Servers, especially when dialing through a remote end FXO port. Please consult your Proxy Server Provider. Calling Features Unconditional Forward: All incoming calls will be forwarded to the “Forwarding Number” automatically.
30 to the FXS port of the gateway and not functioning to FLASH, please adjust the settings in “Flash Detect Time” in category “Advanced Options”. Example of a Three-Way calling: 1. 2. 3. Alex dials to Bob, Bob answers that call. Alex presses Flash and call to Coral (Bob is on hold), Coral answers that call. Alex dials *61 then presses Flash, thus conference call is created. 1. 2. Alex dials to Bob, Bob answers that call.
31 Advanced Options There are two levels to enter Web. Administrator is able to change all settings. Web UI only changes some settings. Web UI auto log out: When logging in a web page, if a user does not act within the effective time range, the user will be disconnected from the web page to allow others to login. Dial Wait Timeout: Use it to set the waiting time for the user’s first key pressing when dialing a number.
32 Enable Out-of-Band DTMF: To send DTMF keys (0~9, *, #,) follow the RFC2833 rules or via SIP Info. Enable Hook Flash Event: The gateway will deliver the flash signal to remote party via RFC2833 or SIP Info. Payload Type：Payload type of RFC2833. Uses Second CPT for VoIP Call: This function is usually applied when the user selects VoIP as the primary path for outgoing calls and PSTN as the backup.
33 Line Settings Listening Volume: Adjusts the hearing volume. Speaking Volume: Adjusts the speaking volume. Tone Volume: Adds a new option to make tone volume adjustable. This setting will be applied to all tones generated by the gateway including Dial Tone, Busy Tone, and so on. Flash Time: FXS: Used to adjust the detecting period of flash signal from the phone set connected to the FXS port.
34 Jitter Buffer: Adjusts the jitter to receive a packet. If the jitter range is too wide, it will delay voice transmission. Silence Suppression: If one side of a connection is not speaking, the system will stop sending voice data (package) to decrease bandwidth usage. Echo Canceling: Prevents poor telecommunication quality caused by echo interference. Packet Time: Defines how long the DVG-3016S send a RTP packet-voice packet- to the other side. The smaller the value, the more bandwidth usage.
35 Digit Map There are 50 sets of leading digit entries to choose voice routing interface – Auto select, PSTN or VoIP. Default Call Route: The default call route can be Auto, VoIP, PSTN and Deny. Auto (VoIP first): The call route is VoIP first, and the next is PSTN. VoIP: The call route is VoIP only. PSTN: The call route is PSTN only. Deny: The call will be deny if the dial-out number is not in the table. Enable: Enable detection of this entry.
36 Speed Dial This system can set up 100 numbers for speed dialing. Setting methods are as follows: Method 1- Single mapping: Fill a short code into the “Speed Dial Code” column, and enter the desired phone number into the “Number To Dial” column. For example, pick up the handset and dial 55＃ and the system will dial 32568791. Method 2- Multi mapping; Fill the prefix code into the ” Speed Dial Code” column and the format to transfer into the “Number To Dial” column.
37 Web: Enable management from Web if ticked. CDR Settings The user can set up a CDR Server to record call detail for every phone call. The present CDR provides the call detail recording in a text file and imports the text file to prepare for an analysis report, if needed. Send record to CDR Server: Enables the call detail recording function. CDR Server IP: Enter the IP address of the CDR server. Port: Enter the listen port of the CDR server.
38 PIN Code: Enter the PIN code (4-6 digits or leave blank. A blank indicates no PIN code is required at this level. Generally, the PIN at level 5 can remain blank to simplify the phone number.) Enable: Enables the PIN code at each level. Privileges: The level is divided into 0~5 (The levels are in descending order; 0 stands for the highest authority and 5 stands for the lowest.
39 Long-Distance Control Table This table controls the level of authority of an outgoing call through FXO. If Level 0 (the highest level) is set to prohibit dialing any number started with prefix 0204, then any level below 0 (including Levels 1 to 5) is also prohibited. If Level 1 is set to prohibit dialing any number with prefix 0, then any level below 1 (including Levels 2 to 5) is also prohibited.
40 Busy Tone Cadence Measurement: Provide a best solution of FXO integrated with PSTN or PBX. FXO will learn the busy tone automatically. BTC Detection Sensitivity: The more sensitivity, the more quickly the system will cut off the call. If the system often cut off an un-finished call, select less sensitivity. CPT parameters Table The CPT has 3 sets of parameter tables. Please adjust the CPT based on local PSTN or PBX.
41 System Information RTP Packet Summary Displays the information of the last finished call. It contains peer IP, peer port, packets sent, packet received and packet lost.
42 STUN Inquiry Ping Test Use “ping” to identify if the remote peer is reachable. Fill in remote IP address and click “Test” will start the test. SNMP Enable SNMP Agent: Enable SNMP if ticked. Get/Set/Trap Community: Enter Community name to Read, Write and Trap. Trap Host: Enter the IP of Trap Host.
43 NTP Time Zone: Set the Time Zone where DVG-3016S resides. Time Server #1~#3: Set the Time Server where DVG-3016S should sync up during start up. (NTP protocol) Backup/Restore You can backup settings to a file and restore settings from that file. You also can restore all settings back to default by selecting Restore Default Configurations and click Restore. Note: It needs to Save Settings and Restart, and all settings will back up default settings or have new setting that you upload.
44 System Operations Save Settings: Save settings after completing. The new settings will take effect after the system is restarted. Please select “Save Settings”. Restart: If it is necessary to restart the system, please select “Restart” and click the “Accept” button. Software Upgrade Gateway provides software upgrade function for a remote end. Software Upgrade Server IP address: Please enter the software server IP address.
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46 5. Coding Principle Instruction After a phone number is entered, dial # to call out immediately or, wait until the “Inter DTMF Timeout” expires (defined in “Advanced Options”, default=4 seconds). If the phone number fits the setting of Digit Map, the gateway dials out the phone number through the assigned interface automatically. The phone number should have at least 2 digits (not including * and #).
47 Start Enter a phone number (D#) Dial the number defined in SpeedDial table Yes Is (D#) defined in Speed Dial table? No Is (D#) defined in Extension table? Yes No Is (D#) defined in Phonebook table? Yes No Is (D#) defined in Phonebook Manager? Yes No Is (D#) defined in SIP proxy server? Yes No Dial (D#) through the first available FXO port to PSTN Yes Does this gateway has an FXO port? No End Dial out as defined in the first match case through the gateway
