Vaid HT-842 User Manual - 2 Installation

Manual is about: ANALOG TERMINAL ADAPTOR

Summary of HT-842

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    User manual analog terminal adaptor model: ht-842 release 1.2 www.Vaidsys.Ru.

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    Release 1.2 1 contents 1 introduction.......................................................................................3 1.1 general information .........................................................................................................3 1.2 protocols ................................

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    Release 1.2 2 4.3.6 discard changes .............................................................................................35 4.4 phone book .............................................................................................................35 4.5 tools...................................

  • Page 4: 1  Introduction

    Release 1.2 3 1 introduction 1.1 general information the is designed as a compact, high performance, and low cost analog terminal adaptor (fxs gateway). It comes with 4 fxs ports that enable traditional analog phone sets or pbx trunk lines to be used for voip communications. The is a full featured f...

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    Release 1.2 4 1.3 hardware specification arm9e processor for high performance dsp for voice codec and voice processing two 10/100m based ethernet ports for wan/lan connections. Led status indicators one fxs port ethernet bridge 1.4 software specification linux os built-in http for accessing internal...

  • Page 6: 2 Installation

    Release 1.2 5 2 installation the device is designed for easy installation and can be installed in various network environments. 2.1 appearance front view front panel there are eight leds on the front panel to display the current status of the 1. Power – this led lights up when the power is applied. ...

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    Release 1.2 6 the has four fxs port (phone), two ethernet ports (lan and pc), power input, and a reset inputs. They are all located at the back panel of the as shown and described below. Back view back panel view 1 ) l4/l3/l2/l1 - these ports are called fxs ports. They should be connected to traditi...

  • Page 8: 2.2  Connection Diagram

    Release 1.2 7 2.2 connection diagram the diagram below shows a typical installation of the device. Four traditional pstn telephones are connected to the four fxs ports. The lan port is connected to a xdsl/cable modem, router or gateway for connection to external network. The pc port is optional and ...

  • Page 9: 3  Basic Operations

    Release 1.2 8 3 basic operations 3.1 star commands the star commands are used to access / control the device via the phone set connected to the phone port. There are two sets of star commands and they are for: 1. Phone configuration 2. Phone operation the first digit of a star command must be “*” an...

  • Page 10: 3.2  Set Up

    Release 1.2 9 pc port ip = 192.168.5.1 (facotry default is bridge mode.) *11983185922# reset all system parameters back to factory defaults. Please see section 2.1 for hardware reset option. Note: these commands are factory preset and cannot be modified. 3.1.2 star commands for phone operation the a...

  • Page 11: 3.3  Phone Operation

    Release 1.2 10 3.3 phone operation the device supports voip calls once it is properly setup for voip service. Phone calls are made via the phone set connected to the phone port. 3.3.1 make a call to make a voip call, place the phone set off hook to hear the dial tone. The dial tone is programmable i...

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    Release 1.2 11 3.3.5 transfer a call (for sip only) if two parties ( a and b) are in an active call with each other. Part a can transfer the call to party c by dialing the star command (the default is *41). The following two transfer modes are supported. A) attended transfer party a dials *41 to hol...

  • Page 13: 4  Web Configuration

    Release 1.2 12 the procedures to activate phone book dialing are: 1. Place the phone off hook 2. Wait for a dial tone 3. Dial the star command (the default is *50), the location (0 to 20), “#” 4. The number in the phone book will be dialed out automatically. 4 web configuration other than auto provi...

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    Release 1.2 13 use pc ip address when: 1. Pc and the pc port (ht-912) are connected and assigned to the same network segment. To access the built-in web server, type the proper ip address (for example: 192.168.2.134 or http://192.168.2.134 in a web browser as shown below (ie, firefox, etc.). Once th...

  • Page 15: 4.2 Status

    Release 1.2 14 once the login is successful, the web browser brings up the status page as shown below. 4.2 status the status page provides a brief summary of the current phone (device) and network information. 4.2.1 phone information 1 serial number each device is assigned with a unique serial numbe...

  • Page 16: 4.3  Configurations

    Release 1.2 15 4.2.2 network information 1 lan port this field shows ip address assigned to the lan port. 2 pc port this field shows ip address assigned to the pc port. 3 pppoe this field shows the dial up status when pppoe is enabled for adsl login. 4 default route the default route shows the ip ad...

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    Release 1.2 16 to change the display language immediately, you can select the language icon as shown below. However, this does not change the default language. 2 time zone – this parameter specifies your local time zone in order for the date/time to be correctly displayed since the date/time obtaine...

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    Release 1.2 17 call request after the last number dialed. The range is 3 to 15 seconds and the default is 5 seconds. 8 network tones – this parameter defines the network tones to be used. The predefined networks tones are: china, hong kong, taiwan, new zealand, united kingdom, united states, korea, ...

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    Release 1.2 18 4.3.2 network configuration this page configures the network interface for lan port and pc port. Lan port – the lan port is intended for internet access. It is normally connected to a network device (router or adsl modem) which has internet access. The following 3 modes are available ...

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    Release 1.2 19 3. Pppoe – this selection is intended for broadband connection (adsl / cable modem) that requires dial up / authentication using pppoe protocol. Both user name and password are required. Please consult your service provider for more information if needed. One advantage with the pppoe ...

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    Release 1.2 20 4.3.3 call settings this page configures all related settings for voip service. Based on the two protocols (h.323 and sip) supported, the operation of device is divided as two endpoint types: h.323 phone and sip phone. Some of the parameters are unique to the endpoint type and are des...

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    Release 1.2 21 the basic h.323 settings are: 1. Endpoint mode – gatekeeper mode supports making a voip call via a call server. Registration to the server is required. Direct mode supports making a voip call by dialing the ip addresses or an alias. 2. Config mode – the device supports two modes: sing...

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    Release 1.2 22 in order to activate the fxs line to be used, the following parameter must be checked for each group. 3. Phone number - this parameter assigns the phone number used for registration in gatekeeper mode. This is used as an alias in direct mode. 4. Display name – this parameter (optional...

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    Release 1.2 23 function properly or reliably advanced settings more settings are available under the advanced settings tab. These setting s are common to all h.323 configurations. Depending on your network requirements, please consult your network administrator for the correct configuration. 1. Ras ...

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    Release 1.2 24 registration message. Register multiple times mode means that each number is registered in a separate registration message. 6. Dtmf signaling – this parameter sets the method of sending dtmf signals. Inband measns that the dtmf signal is sent as an analog signal via the voice channel....

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    Release 1.2 25 relay proxy mode is a proprietary nat protocol and it requires the use of our relay proxy server. All voip signaling packets are encapsulated (encrypted for more secured transmission if enabled) and transmitted via another port/channel. Up to 4 backup relay servers are supported. Once...

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    Release 1.2 26 in sip phone mode, it supports two registration modes: single server and config by line. In single server mode, only one sip registration is used for all 4 fxs lines. The will perform line hunting automatically when a call is made or a call comes in. In config by line mode, each fxs l...

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    Release 1.2 27 d) no answer forward – call is forwarded when it is not answered. E) busy or no answer forward – call is forwarded when the line is in use or not answered. 13. Forward number – this defines the number to be used for call forward. 14. Backup server – the backup option provides settings...

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    Release 1.2 28 1. Signaling port – this port is used to convey signaling message with the sip proxy. The standard port number is 5060. 2. Nat keep-alive – when enabled, a dummy packet i sent to the local firewall / router in order to keep the ports opened for voip service. 3. P2p – this enables peer...

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    Release 1.2 29 d) avs – this encryption is developed by a network equipment vendor in shanghai, china. E) et263 – this encryption is supported by major network equipment vendors in china. 8. Signaling nat traversal – nat traversal is an algorithm designed to solve a common problem in tcp/ip networki...

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    Release 1.2 30 1. Rtp port (range) – audio stream is transmitted via real time protocol (rtp) and at least 4 ports are used per voice channel. The default port range is 16384 – 32768. Specify the port range depending on your network environment if needed. 2. Packet length (ms) – this specify the len...

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    Release 1.2 31 8. Relay proxy mode is a proprietary nat protocol and it requires the use of our relay proxy server. All voip signaling packets are encapsulated (encrypted for more secured transmissions if enabled) and transmitted via another port/channel. Three relay modes of operation are supported...

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    Release 1.2 32 4.3.3.4 dial plan dial plan defines how the device processes a number is dialed. This field is located in the calling setting window and it is available for both h.323 and sip modes. The dial plan is very flexible and can be configured for a wide range of dialing applications. The bas...

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    Release 1.2 33 3. Dial plan = “001:-001+1751” means that the number dialed out will the first 3 digits “001” changed to “1751” when a number with the first digits as “001” is entered. A) number entered = “00182121234”, actual number dialed = “175282121234”. B) number entered = “82121234”, actual num...

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    Release 1.2 34 1. Star commands – star commands for phonebook function (default is *50) is programmable in this page. The format is “*”+digits and the number of digits is not limited. However, the detection is based on the first match. For example, if “*1” is defined, then the definition of “*1x..” ...

  • Page 36: 4.4 Phone

    Release 1.2 35 4.3.5 save changes when all changes have been made, click on the save changes tab to save all settings to the flash memory. The message window below is displayed when the saving is completed. 4.3.6 discard changes click on the discard changes tab to ignore all changes made. 4.4 phone ...

  • Page 37: 4.5 Tools

    Release 1.2 36 4.5 tools the tools section is intended to offer the following functions: online upgrade, change password, reset config, and reboot. 4.5.1 online upgrade click on the online upgrade tab to perform manual firmware upgrade. Enter the upgrade address as shown below. Please contact your s...

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    Release 1.2 37 warning: once the upgrade starts, a message window is display to show the upgrade status. Do not turn off the power while the firmware upgrade is in process! 4.5.2 change password the device supports two login levels to the built-in webpage. The user level is intended for general user...

  • Page 39: 4.6 Gain Settings

    Release 1.2 38 please also see section 3.1.1 for a star command reset option. 4.5.4 reboot click on the reboot tab to reboot the device. 4.6 gain settings this gain settings page is hidden and is only intended for users who is really interested in adjusting the receive level (input gain) and transmi...

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    Release 1.2 39 www.Vaidsys.Ru ht-842 user manual.