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2. k hone SERI IP Phone Terminal Gain Settings Line 1 Line 1 Output Gain DdB Line 1 Input Gain x Line 2 Line 2 Output Gain ng Line 2 Input Gain OdB 3 Line 3 Output Gain DdB Line 3 Input Gain OdB Line 4 Line 4 Output Gain Line 4 Input Gain amp uB Release 1 2 39
3. 25 ASS Medid ua de NM EM D M EE 20 nus LI 32 PHONE Setn SOOT 33 4 35 59 35 Release 1 2 1 HT 842R User Manual 2520 Discard G7 ae 35 4 4 OMG eee a unu car se 35 4 5 36 2521 MOOI TET 36 45 2 Change AS SW OM e as tu UE ERU 37 453 LH D 37 RODO R 38 APSO CTS det ou Rc diste INE 38 Release 1 2 2 HT 842R User Manual 1 Introduction 1 1 General Information The HT 842R 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 HT 842R is a full featured FXS gateway and is designed for easy installation and configuration It supports the two most widely used Open VoIP Standards SIP and H 323 This allows the HT 842R interoperate seamlessly with softswitches or IP PBXs made by various vendors Its high performance offers toll quality voice flexible networking and f
4. On Off 4000 a On Off 5 Caller ID This defines the Caller ID signal to be sent from the FXS port It supports both Bellcore Telcordia and ETSI FSK Caller ID standards The CID FSK Mode sets up the FSK modem to be used Bellcore for Bell 202 and ETSI for V 23 Please refer to the Caller ID standard desired for further information CID FSK Mode Bellcare CID Signaling 1st Ring Duration 1000 Delay Time Before CID Delay Time After CID 1000 1 6 Hot Line When this feature is enabled the Hot Line Number defined will be dialed out automatically whenever the phone 15 off hook Release 1 2 34 HT 842R User Manual Hot Line Line1 Hot Line Enable C Disable 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 Configurations Preference Network Call Settings Phone Settings save Changes Discard Changes The message window below 15 displayed when the saving is completed http 197 168 7 744 Configuration saved 4 3 6 Discard Changes Click on the Discard Changes tab to ignore all changes made 4 4 Phone Book This page lists all the phone book entries for editing Just click on Save Changes to save all entries to memory The Name field is optional only the Number field is used for memory dialing Release 1 2 35 HT 842R User Manual E IP Phone Terminal Ed Phone
5. 3 Dial Plan 2 001 001 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 number dialed 82121234 4 Dial Plan means that the input number is limited to 4 digit long and will be dialed out immediately when the fourth digit is entered Dial Plan 0 means that the input number is restricted to 11 digit long and the first two digits must be 13 When this condition is matched the number dialed out will have a leading 0 added a Number entered 13901234567 actual number dialed 013901234567 b Number entered 12801234567 actual number dialed 12801234567 6 Dial 13 6 0 means that the input number is restricted to 11 digit long and the first two digits must be 13 and the third digit can be 6 8 or 9 When this condition is matched the number dialed out will have a leading O added a Number entered 013971234567 b Number entered 13001234567 actual number dialed 13001234567 N 13901234567 actual number dialed Please note that the above samples are simple and intended to show the meaning of various rules They may not have any practical m
6. DC24V 300mA c One Ethernet cable 3 Meter long Release 1 2 4 HT 842R User Manual 2 Installation The device is designed for easy installation and can be installed in various network environments 2 1 Appearance HT 842R Front View Easy Gateway O O O O POWER RUN LAN PC L1 L2 L3 L4 MODEL NO HT 842R Made in China HT 842R Front Panel There are eight LEDs on the front panel to display the current status of the HT 842R 1 Power This LED lights up when the power is applied 2 RUN This LED flashes at a rate of 100ms ON and 100ms OFF when the device is booting up or connecting servers It flashes at a rate of 1s ON and 1s OFF when server connection is established It does not flash or turn on when the device fails LAN This LED shows the LAN port status It flashes when link and data activities occurs 4 PC This LED shows the PC port status flashes when link and data activities occurs 5 L1 L2 L3 LA These LEDs show the hook status of the corresponding FXS ports Each LED lights up when the corresponding FXS line is OFF hook state In Use Release 1 2 5 HT 842R User Manual The HT 842R has four FXS port Phone two Ethernet ports LAN and PC Power Input and a Reset inputs They located at the back panel of the HT 842R as shown and described below HT 842R Back View L2 L1 DC24V 500mA RESET HT 842R Back Panel View 1 L4 L3 L2 L1
7. information 4 2 1 Phone Information 1 Serial Number Each device is assigned with a unique serial number by the factory This number is important for auto provision technical support and warranty repair The product label at the bottom also contains this information 2 Firmware Version This field identifies the current Firmware Version installed 3 Hardware Model This field identifies the hardware model and version 4 Phone Status This field shows the status of server registration for each FXS port 1f the device registers to the designated server s successfully it displays the status LOGIN Otherwise it displays LOGOUT Release 1 2 14 HT 842R User Manual 4 2 2 Network Information 1 LAN Port This field shows IP address assigned to the LAN port 2 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 address of the default gateway router that is used in the current network environment 5 DNSServer This field shows the IP address of the DNS server to be used for domain name interpretation 4 3 Configurations To access the Configurations page click on the Configurations tab on the left hand column This brings up all the pages under this tab Preference Network Call Settings and Phone Settings 4 3 1 Preference This page configures the general settings in the devic
8. Book fd Save Changes 9 Tools Status Configurations 4 5 Tools The Tools section is intended to offer the following functions Online Upgrade Change Password Reset Config and Reboot Tools Online Upgrade Change Password Reset Config 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 service provider to determine if there is a new firmware available Release 1 2 36 HT 842R User Manual Online Upgrade Current Version A38H5 3 15 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 and is restricted from accessing the Call Settings page and Reset Configuration function In this level only the password for the user level can be changed The default password for the user level login ID user is 1234 The Administrator level allows full accessing to the DEVICE configurations In this level the password for both levels can be change The default password for the administrator level login ID admin is admin It is important to record the new password s lf the admin password is lost a special star command is avail
9. Single Server and Config by Line In Single Server mode only one SIP registration is used for all 4 FXS lines The HT 842R will perform line hunting automatically when a call is made or a call comes in In Config by Line mode each FXS line has its own SIP registration and they are treated as independent lines A Backup Server option is also available for each FXS line to insure a more reliable SIP Service The basic SIP settings are 1 Phone Number This parameter assigns the phone number used for SIP registration 2 Phone Number 2 This parameter assigns the second phone number used for registration This acts as a second phone number for SIP calls 4 Display Name This parameter optional specifies the Caller name and is transmitted as part of the caller ID 5 SIP Proxy A SIP Proxy acts as a call manager of all incoming and outgoing calls Specify the location IP address domain name of the designated SIP Proxy used for SIP service The standard port used is 5060 To specify a non standard signaling port add lt port number to the of the location For example If SIP Proxy yousippbx com the signaling port is the standard port 5060 If SIP Proxy yoursippbx com 15060 the signaling port is 15060 6 SIP Registrar A SIP Registrar maintains a database of all SIP phones registered and their contact information Specify the location IP address domain name of the designated SIP Registrar The standard port used is
10. 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 Slovenia Czechoslovakia India Singapore Israel Malaysia Indonesia Thailand Romania Bangladesh and Customized The Customized option allows user to define his own network tones If the desired network tones selection is not available user can use this Customized option Network Tones Dial Tone fo Ring Back Tone Indication Tone fo Each network tone contains 16 parameters as shown below number of cadences repeat counter 0 infinite 1 n repeat 1 n times cadence one on in milliseconds cadence one off in milliseconds cadence two on in milliseconds cadence two off in milliseconds cadence three on in milliseconds 1 SEM xt cadence three off in milliseconds 9 tone 1 frequency 300 3000 Hz 10 tone 2 frequency 300 3000 Hz 11 tone 3 frequency 300 3000 Hz 12 tone 4 frequency 300 3000 Hz 13 tone 1 level 0 31 023dB 1dB step 14 tone 2 level 0 31 023dB 1dB step 15 tone 3 level 0 31 0 3dB 1dB per step 16 tone 4 level 0 31 0 3 1dB per step Below are two sample network tone definitions for reference 1 A New Zealand Dial Tone 400 Hz is defined as 0 0 0 0 0 0 0 0 400 0 0 0 10 0 0 0 2 A New Zealand Busy tone 400Hz with a cadence of 500ms on and 500ms off repe
11. 2 168 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 available star commands for phone operation are 3 digit long at least and some contains operands These Star Cmmands are shown in the table below and they can be programmed to other values via the built in Web Server Star Command Function 42 the current call Release the Hold cal ILE Call Transfer to another VoIP Number 3 2 Setup The device supports two major configuration methods 1 Provisioning Server The device can be programmed at the factory via a DHCP Host or via the built in web server to execute an Auto Provisioning Procedure to obtain a configuration file from a Provisioning Server a HTTP or a TFTP server This configuration file contains all the necessary parameters to set up the device for VoIP Services This method requires no manual operation once the Auto Provision mode is set and greatly simplifies the installation and configuration of the device This is a proprietary method Please your local support for more information 2 HTTP Web Server The device comes with a built in HTTP Web Server for user configuration on the same network segment can access the built in Web Server by entering the IP address in a
12. 2R User Manual 4 33 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 described separately below Call Settings Endpoint Type H 323 Phone Advanced Settings Endpoint Mode Gatekeeper Made Media Settings Contig Mode Single config Phone Number Display Name H 323 ID ll Gatekeeper Address Enable vOSI AVS Encryption Enable Authentication Dial Plan Fax 4 3 3 1 323 Phone The 323 Phone selection for Endpoint Type refers the protocol used Call Settings Endpoint Type Endpoint Mode Gatekeeper Config Mode ConfigbyLine gt Line1 2 Line3 Line4 Phone Number H 323 ID 1421 Gatekeeper Address Enable VOS AVS Encryption 4 235 Auth Fax Line Release 1 2 20 HT 842R User Manual 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 Endpoint Type H 323 Phone we Endpoint Mode Direct Made Direct Mode Gatekeeper Made 2 Config Mode The device supports two modes Single Config Config by Line and Con
13. 5060 To specify a non standard signaling port add lt port number gt to the of the location For example If SIP Proxy yousippbx com the signaling port is the standard port 5060 If SIP Proxy yoursippbx com 15060 the signaling port is 15060 7 Registry Expiry s This specifies the expiry duration at the SIP Registrar after a successful registration The range is 60 to 36400 seconds 8 Outbound Proxy A network node acts as proxy for outbound traffic between a client and a server Please contact your network administration to determine if this proxy is available or not 9 Home Domain This field enables the use of home domain name is SIP registration instead of IP address 10 Authentication ID This field specifies the ID to be used for Authentication during a SIP registration 11 Password This field specifies the password used for Authentication during a SIP registration 12 Call Forward Type This defines the Call Forward condition and the available options are a Not Forward Call forward is disabled b Unconditional Forward Call is always forwarded c Busy Forward Call is forwarded when the line is in use engaged Release 1 2 26 HT 842R User Manual No Answer Forward Call is forwarded when it is not answered Busy or No Answer Forward Call is forwarded when the line is in use or not answered Call Forward Type Not Forward or Forward Unco
14. DTMF signals Inband measns that the DTMF signal is sent as an analog signal via the voice channel Outband means that the DTMF signal is sent as DTMF command the data channel and is commonly known as RFC2833 In Outband mode payload type is required and the default type is set to 101 DTMF Signaling Outtrand Y 7 Signaling QoS This parameter sets the QoS mode for VoIP Signaling for better response time and more reliable VoIP Call signaling Both IP TOS and Diffserv modes are supported Please check with your network administrator or ISP for the correct setting Signaling 20 8 Signaling NAT Traversal NAT Traversal is an algorithm designed to solve a common problem in TCP IP networking in establishing connections between hosts in private TCP IP networks that use NAT devices This parameter only sets the NAT Traversal mode for VoIP signaling The 3 methods supported are Citron Port forward DMZ and Relay Proxy Both NAT Citron and Port forward DMZ are well known NAT protocols are are widely used however they require the support of local network Relay Proxy mode is a proprietary NAT protocol and it is designed for NAT Traversal with the capability of avoiding VoIP blockings All VoIP signaling and or media packets are encapsulated encrypted as well if enabled and transmitted via another port channel to our proprietary Relay Server Please contact your service provider to determin
15. These ports are called FXS ports They should be connected to traditional PSTN phones or PBX trunk lines via standard telephone cables Please DO NOT connect these ports to CO Lines or PSTN PBX Extension Lines since this may cause damages to the device 2 LAN This port is to be connected to a network device for internet network access 3 PC This port is to be connected to a PC laptop or a network device 4 POWER DC24V 300mA The AC DC Adapter provided is connected to this port 5 Reset The device is reset and rebooted when this push button switch is pressed Release 1 2 6 HT 842R User Manual 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 it is intended to build a new local network segment Fixed IP mode or extend the existing network segment Bridge mode Once the wiring is done the AC DC Adapter should then be connected to power up the device 1 AC DC Power Adapter Use the one Provided or 7 equivalent F Local Network Connection External Network PC other Network Connection Device xDSL Cable Modem Router Gateway etc HT 842R Connection Diagram Release 1 2 7 HT 842R User Manual 3 Basic Operations 3 1 Star Commands The star command
16. User Manual ANALOG TERMINAL ADAPTOR Model HT 842R Release 1 2 HT 842R User Manual Contents 1 14 General rire siii ID 3 eae ib E ER 3 Lo Hardware duod deed aun Lara iud 4 SOM WATS 4 1 5 Listof ierit rari 4 2 JlInstallallOn Paese on pa 5 PAPE Gottes RIO TRIP Tur Uc 2 2 Basic 8 0 Jl OU NU M 9 3 1 1 Star Commands for Phone Configuration 8 3 1 2 Star Commands for Phone Operation 9 ae UU PN D a Sona eae 9 3 9 JPhole o totales dolls 10 Makea da ius UMS es ME 10 332 AnsWera Gall boa bises pbi a eR b esM EE LE 10 3 3 3 Answer a Call Waiting Call not available in this model
17. Web Browser Please refer to Section 4 for more detailed information Release 1 2 9 HT 842R User Manual 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 in the Preference page of the built in web server Justdiala valid VoIP number and then to make a call lf the digit is omitted the call will be dialed out when the Auto dial Timeout expires This timeout is programmable in Preference page and the default is 5 seconds Please note that the phone set must be set to DTMF dialing mode 3 3 2 Answer a Call When an incoming call occurs the phone set will ring at a ringing pattern defined in the Phone Setting page Just place the phone set off hook to answer the call 3 3 8 Answer a Call Waiting Call not available in this model When an incoming call occurs during an active call it is referred as a Call Waiting Call The user hears an alerting CAS tone when a Call Waiting call occurs This call waiting feature is only available in SIP H 323 does not support this feature Press the FLASH key on phone keypad or flash the hook switch to answer a Call Waiting Call 3 3 4 Hold a Call For SIP only Dial the Star Command the default is 42 to put the active call on hold To release t
18. able to reset all system settings Please refer to section 3 1 1 for more information User Level Administration Level Confirm Password Change 4 5 3 Reset Configuration This function can only be accessed in administrator login level Click on the Reset Configuration tab to initiate the reset process message windows pops up to ask for confirmation Click Yes to reset all configurations back factory defaults Click No to cancel Once the reset process is completed the device reboots itself Release 1 2 37 HT 842R User Manual Tools Online Upgrade Change Password Reset Config Heboot 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 transmit level Output Gain of each FXS port The URL of this pge is htto xxx xxx xxx xxx en_ US gain html Enter this filed a web browser and the GAIN SETTINGS page pops up If you have not logged in to the web server you will need to login first The range of the gain setting is from 36 dB to 36 dB Please adjust the gains with caution If the input gain is too high the DTMF dialing may not be detected properly You can always click on Reset to resume to system default level Release 1 2 38 HT 842R User Manual fer
19. at is defined as 1 0 500 500 0 0 0 0 400 0 0 0 10 0 0 0 Release 1 2 17 HT 842R User Manual 4 3 2 Network Configuration This page configures the network interface for LAN Port and PC Port Network Configuration LAN Port PPPoE PC Port Static IP v 802 1q VLAN 9 Enable Disable Advances Advance gt gt LAN Port The LAN port is intended for internet access Itis normally connected to a network device router or ADSL modem which has internet access The following 3 modes are available for selection Network Configuration LAN Port Username Password 802 1q VLAN 9 Enable Disable VLAN i nena vanos Advance lt ci ee Address Address 1 DHCP This mode should be selected If the network device functions a DHCP host This allows the DEVICE to obtain all related network information settings from the DHCP host 2 Static IP This mode sets the LAN port IP manually which can either be a public or private IP Other network settings Subnet Mask Default Route Primary DNS Secondary DNS should also be entered accordingly Network Configuration LAN Port Static IP IP Address y Default Route Primary DNS SS y DN S optional Release 1 2 18 HT 842R User Manual 3 PPPoE This selection is intended for broadband connection ADSL Cable modem that requires dial up authentication using PPP
20. c string Some gatekeepers may use this ID for authentication 6 Gatekeeper Address This assigns the location of the Gatekeeper for VoIP Service 7 VOS AVS Encryption Both VOS2000 AVS Encryption methods are used by major network equipment vendors in China to avoid VoIP blocking in order insure a reliable VoIP services In order to use this your VoIP service provider needs to support this encryption method For H 323 VOS AVS Encryption can be enabled or disabled for each number registration VOS Encryption supports two modes Signaling Encryption and Signaling and Media Encryption Please consult your services provider for more information 8 Authentication If H 235 authentication is required enable this field and enter the H 235 ID and Password 1H 235 Auth 9 Fax Fax function is supported and can be enabled via this field Fax can be transmitted via inband signal G 711 or outband T 38 commands Please note that there are many factors that can affect the fax transmission and it may not Release 1 2 22 HT 842R User Manual function properly or reliably Fax lt lt Line1 Fax Disable 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 Advanced Settings RAS Port Fast Start 9 Enable Disabl
21. date and time information t accepts both domain name and IP address 4 Auto Provision This parameter enables or disables the Auto Provision procedures The Auto Provision is a batch script to obtain configuration and firmware upgrade information from a server Once this option is enabled two additional parameters Provision Server and Provision Interval are displayed The Provision Server specifies the location of the designated provision server The auto provision procedure is executed at boot up time and is repeated at a duration specified in the parameter Provision Interval Auto provision 9 Enable Disable provision Sever 7 provision interval 5 Remote Control This feature is intended for remote technical support and it enables the device to connect to a remote server Please contact your service provider for more information on this feature 6 as Delimiter When dialing a VoIP number the VoIP device needs to wait for the user to complete the number dialing before the call request is actually sent to the server This parameter enables or disables the key to be used to signal the number dialing is completed and the call request can be execute immediately 7 Auto Dial Timeout This parameter sets the maximum delay for sending out a Release 1 2 16 HT 842R User Manual call request after the last number dialed The range is 3 to 15 seconds and the default is 5 seconds 8 Network
22. e Register Mode DTMF Signaling Signaling 005 Media Settings 1 RAS Port This Port is used to convey the registration admissions bandwidth change and status messages between two H 323 endpoints If not specified the port address is assigned automatically 2 0 931 Port This port is used for call signaling to convey Call Setup and teardown messages between two H 323 endpoints If not specified the port address is assigned automatically 3 H 245 Port The H 245 requires at least 2 ports for media control protocol should be specified as a port range If not specified the port address is assigned automatically 4 Fast Start Fast Start is a new method of call setup that bypasses some usual steps in order to make it faster In addition to the speed improvement Fast Start allows the media channels to be operational before the CONNECT message is sent which is a requirement for certain billing procedures Leave this enabled if you are not sure 5 Register Mode Two registration modes are support Register Multiple Numbers mode means that multiple numbers are registered in a single Release 1 2 23 HT 842R User Manual registration message Register Multiple Times mode means that each number is registered in a separate registration message Register Mode Register Multiple Mur Re giste r Multi ple Nurnbe Register Multiple Times 6 Signaling This parameter sets the method of sending
23. e Language Time Zone Time server Auto Provision as Delimiter Auto dial Timeout Network Tones INFO Server China Phone Code Preference Language s v Key as Delimiter 9 Enable Disable Time Zone GMT 8 Auto dial Timeout Time Server panl ntp arg Metwork Tones China Auto provision Q Enable 9 Disahle INFO Server 422 FemoteControl China Phone Code Enable 9 Disable 1 Language his field sets the language to be used for initial access to the built in Web Server The languages currently available for selection are English Simplified Chinese and Traditional Chinese 8 x Once the language change 15 saved it does not take effect until the device is rebooted Release 1 2 15 HT 842R User Manual 2 v English fal ib rp Mr Sete Pat To change the display language immediately you can select the language icon as shown below However this does not change the default language IP Phone Terminal 2 Time Zone This parameter specifies your local time zone in order for the date time to be correctly displayed since the date time obtained from a network time server is referenced to the Greenwich Mean Time GMT If your time zone is 8 hours ahead of the GMT you need to enter the value GMT 8 in this field 3 Time Server This parameter specifies the location of the network time server for obtaining the
24. e if this mode is supported Release 1 2 24 HT 842R User Manual Signaling Jone Traversal Nat Citron RTP Port range Port farward DHMz Relay 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 the designated Relay Server fails the next available Relay Server on the back up list will be used Once the designated Relay Server resumes operation it will be used instead of the back up Relay Server Note For Service providers RELAY Proxy software is available at no charge Please contact your supplier for support For end user please contact your service provider to see if this feature is available 4 3 3 2 SIP Phone The SIP Phone selection for Endpoint Type refers to the SIP protocol used Call Settings Endpoint Type sPPhone H Config Mode single Server Mode Phone Number Phone Number 2 Display Name 55 SIP Proxy E SIP Registrar Server Register Expiry s Outbound Proxy _ Home Domain uL AuthenticationID Dial Plan 14 Call Forward Type NotForward wy Call Fonvard Number asy Backup Server C Enable Disable Fax Line gt Release 1 2 25 HT 842R User Manual In SIP Phone mode it supports two registration modes
25. eaning A combination of these rules joined with the symbol can be realized for a much more complicated dialing application 4 3 4 Phone Settings The Phone Settings page configures the FXS port and its related operations They are described in details below Release 1 2 33 HT 842R User Manual Phone Settings PhoneBook Function FXS 48v Standby C Enable Disable Billing Support Enable Disable Ring Parameters Hot Line 1 Star Commands Star Commands for PhoneBook Function Default is 50 is programmable in this page The format 15 1015 and the number of digits 15 not limited However the detection is based on the first match For example if 1 is defined then the definition of 1x is possible but will not be detected as a star command Star command for Call Transfer and Call Hold are preset by the factory and cannot be changed Please refer to Section 3 for more information 2 FXS 48V Standby Normal FXS on hook line voltage is 24V Enable this option to change the on hook line voltage to 48V 3 Billing Support Enable this option to enable call records to be sent to our Billing Server for billing purpose Please contact your provider for more information on this 4 Ring Parameters The ringing frequency and cadence at the FXS port can be set here It allows settings of 3 ringing cadences per ring cycle Ring Parameters lt Ring Frequency Ri ng Cadence One 1200
26. eature rich call functions It is an ideal low cost solution for SME environment where multiple lines are required 1 2 Protocols TCP IP V4 IP V6 auto adapt ITU T H 323 V4 Standard H 2250 V4 Standard H 245 V7 Standard H 235 Standard MD5 5 1 ITU T G 711 Alaw ULaw G 729A G 729AB and G 723 1 Voice Codec HFC1889 Real Time Data Transmission Proprietary Firewall Pass Through Technology SIP V2 0 Standard oimple Traversal of UDP over NAT STUN Web base Management PPP over Ethernet PPPoE PPP Authentication Protocol PAP Internet Control Message Protocol ICMP TFTP Client Hyper Text Transfer Protocol HTTP Dynamic Host Configuration Protocol DHCP Domain Name System DNS User account authentication using MD5 Out band DTMF Relay RFC 2833 and SIP Info Release 1 2 3 HT 842R User Manual 1 8 Hardware Specification 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 parameters PPPoE dial up Network Address Traversal NAT and Router functions DHCP Client DHCP Server Firmware On line upgrade Phone Book Memory Dial Caller ID Multiple Language Support Billing Information for accounting purpose 1 5 List of the Package a One HT 912 FXS Gateway b One AC DC Adapter
27. er This feature requires the support from the VoIP service provider and is enabled in the built in web server 3 3 7 Hot Line The Hot Line feature sets the device to dial a preset VolP number whenever the phone goes off hook No other numbers can be dialed This feature is enabled via the built in web server 3 3 7 Phone Book The Phone Book feature offers 20 entries of names and phone numbers Users can access the Phone Book via the built in Web Server There are two fields for each entry Name and Number The Name field is optional and used for reference to the number entered The Number field is used for Phone Book Dialing and its value can be a number or a name If this field is empty the entry is considered as empty Release 1 2 11 HT 842R User Manual HO Name Number 01 jonsoon 02 03 tony 11 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 4 The Number in the Phone Book will be dialed out automatically 4 Web Configuration Other than Auto Provisioning the device comes with a built in Web Server HTML for the device configuration In order to access this Web Server the LAN or PC IP Address must be known Star Commands are available to generate a voice prompt of the LAN or PC IP Address required and to assign IP Address to both LAN and PC ports Please refer to Section 3 1 1
28. erformance and operation of the Media channel Media Setings Port Range 6384 52768 PacketLengthims Jitter Buffer Delay ms Media QoS Media Encryption iw Symmetric RTP Media Traversal None Release 1 2 29 HT 842R User Manual 1 RTP Port range Audio stream is transmitted 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 length of a voice packet The default packet length is 20 ms 3 Jitter Buffer Mode Three jitter modes are available The Fixed Mode which is the default mode is a simple first in first out mode with a fixed jitter buffer delay By definition the jitter buffer depth is twice the jitter buffer delay The Sequential Mode is also a fixed jitter buffer delay mode but in this mode the jitter buffer function looks at the packet timestamp for dropped or out of sequence packet problems The data packets are sorted based on the packet timestamp The Adaptive Mode optimizes the size of the jitter buffer delay and depth in response to network conditions in addition to the sequential mode 4 Media QoS QoS is also available for Media packets to improve voice quality This is rather significant in a network environment with large amount of data traffics Both IP TOS and DiffServ methods are su
29. fails the next available Relay Server on the back up list will be used Once the designated Relay Server resumes operation it will be used instead of the back up Relay Server Media NAT Traversal Address 14222 User Name 1422 Passworil 14222 Encryption Relay Mode Backup Relay Server Relay Server Relay Server Relay Server 4 Note For Service providers RELAY Proxy software is available no charge Please contact your supplier for support For end user please contact your service provider to see if this feature 15 available 9 Audio Codec Preference The table below list the voice codec priorities in descending order Each voice codec can be enabled place a check mark in the check box or disabled individually Select the voice code and then click on the UP or DOWN button to move the order on the list Release 1 2 31 HT 842R User Manual Audio Codec lt lt alaw DOWN ulaw 9729 0729 g729ab g7231 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 basic syntax is lt event gt lt action gt lt event gt lt action gt where event defines the event to be matched event consists of a
30. fig by Group Singel Config allows only one phone number and Gatekeeper configuration Config Mode Single Config Phone Number 14442 Display 1422 Gatekeeper Address Config by Line allows each FXS line to have its own phone number H 323 ID and Gatekeeper configuration Different FXS line can register to the same or different Gatekeeper Config Mode ConigbyLine gt Line1 CLine2 CLine3 C Line 4 Phone Humber batekeeper Address 142220 Enable VOS AVS Encryption 1H 235 Auth Config by Group allows up to 4 groups of phone number H 323 ID and Gatekeeper configuration Each FXS line can be assigned to each group This creates a very flexible mappings for multiple Gatekeepers to share the 4 FXS lines Release 1 2 21 HT 842R User Manual Config Mode Config by Group PhoneNumber H 323 ID Gatekeeper Address Enable VOS AVS Encryptian 235 Auth Activated Lines in Group 2 11 112 114 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 specifies the Caller name and is transmitted as part of the caller ID 5 323 ID This parameter is specified in the H 323 protocol It is an identifier containing an alphanumeri
31. for more information 4 1 Access the Built in Web Server The built in Web Server can be accessed by typing the LAN PC IP address in PC web browser Please see below to determine which IP Address to be used to access the built in Web Server Use LAN IP address when 1 PC and the LAN port are connected and assigned to the same network segment This condition applies to the PC that is connected to the PC port with bridge mode set 2 LAN IP address is public and the PC has internet access Note If a private IP is assigned to the LAN port it may still be accessed from the internet provided that the local router is set up properly Please consult your network administrator for more information Release 1 2 12 HT 842R User Manual 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 Web Browser as shown below IE Firefox etc MSN Hong Kong Homepage Windows Internet Explorer ETT ET P Ive http 192 168 2 134 Once the device responds to the HTTP request the Web Browser will prompt for a login window as shown below Connect to 197 166 7 134 The server 192 158 2 134 at Please Login requires Username and password Warning This server is requesting that your username and password be sent in an insecure ma
32. he call on hold dial the Star Command again the default is 42 place the phone off hook if it is already on hook place the phone on hook and then off hold if the phone is still off hook press the flash key if the phone is still off hook This feature applies to VoIP SIP calls only Release 1 2 10 HT 842R User Manual 3 3 5 Transfer a Call For SIP only If two parties A and B are 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 hold the call with Party B and then dials the phone number to call Party C after hearing a dial tone After Party C answers the call Party A can then hang up to complete the transfer Party B and Party C will then be connected b Unattended Transfer Party A dials the Star Command the default is 41 to hold the call with Party B and then dials the phone number to call Party C after hearing a dial tone Party can then hang up the call when hearing a ring back tone If Party C answers the call Party C will then be connected with Party B 1f Party C does not answer Party A s phone will then ring after the call to Party C is terminated 3 3 6 Call Forward For SIP only The Call Forward feature allows a call to be forwarded to a designated number under the following conditions Unconditional Busy No Answer Busy or No Answ
33. his 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 networking in establishing connections between hosts in private TCP IP networks that use NAT devices This parameter only sets the Traversal mode for VoIP signaling The 2 methods supported are STUN RFC3489 and Relay Proxy A STUN Server is required for STUN RFC3489 Signaling NAT Traversal STUN RFC 3489 Relay Relay Proxy mode is a proprietary NAT protocol and it requires the use of our Relay Proxy Server All VolP 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 the designated Relay Server fails the next available Relay Server on the back up list will be used Once the designated Relay Server resumes operation it will be used instead of the back up Relay Server Note For Service providers RELAY Proxy software is available at no charge Please contact your supplier for support For end user please contact your service provider to see if this feature is available 4 3 3 3 Media Settings Once a VolP call is established the Media channel is used for voice transmission The settings listed below configure the p
34. nditional Forward Backup Server Busy Forward Answer Forward Busy Ma Answer Forw 13 Forward Number This defines the number to be used for Call Forward 14 Backup Server The backup option provides settings for a SIP backup server Once the designated SIP Proxy and SIP Registrar fail the backups will be used automatically 15 Fax Fax function is supported and can be enabled via this field can be transmitted via inband signal G 711 or outband T 38 commands Please note that there are many factors that can affect the fax transmission and it may not function properly or reliably Fax Linez Line1 Fax Disable Advanced Settings More settings are available under the Advanced Settings tab Depending on your network requirements please consult your network administrator for the correct configuration Release 1 2 Advanced Settingss Signaling Port NAT Keep alive 6 Enable Disable P2P C Enable Disable Virtual Ringback C Enable Disable Advanced Timing DTMF Signaling Outband DTMF type RTP Payload Type Signaling 20 Signaling Encryption akate Traversal 27 HT 842R User Manual 1 Signaling Port This Port is used to convey signaling message with the SIP Proxy The standard port number is 5060 2 Keep alive When enabled a dummy packet sent to the local firewall router in order to keep the ports opened for VoIP service 3 P2P This enables Peer
35. nner basic authentication without a secure connection User name wv Password Remember my password The device supports two login levels For Administrator please enter User name admin and Password admin factory default For User please enter User name user and the Password 1234 factory default Both passwords can be changed in the Administrator mode Only user password can be changed in the User mode Please keep a record of the new passwords if changed There is a otar Command to reset the passwords to the factory defaults The Administrator mode allows full access to the built in Web Server whereas the User mode restricts the user from accessing the Call Settings page Release 1 2 13 HT 842R User Manual the login is successful the Web Browser brings up the Status page as shown below fer IP Phone Terminal Status Status Phone Information Network Information Configurations Number HT842R48I08059038 LAH Port 192 168 2 122 Firmware Version A38HS 3 15 MAC 00 11T BE 02 13 BE Phone Book Hardware Model Afs PC Port In Bridge Mode Line Register 1 TOM ih LOGIM PPPoE Disabled Line Register 2 LOGIN Default Route 132 168 2 1 Status Line Register 3 mm DNS Server 202 130 97 65 Status ee Line Register 4 Status ro 4 2 Status The Status page provides a brief summary of the Current Phone Device and Network
36. oE protocol Both User Name and Password are required Please consult your service provider for more information if needed One advantage with the PPPoE dial up is that the IP address obtained for the LAN port is normally a public IP Network Configuration LAN Port PPPoE v User name Password More advanced parameters for 802 14 VLAN and MAC settings available Please consult your network administrator for assistance if needed PC PORT The PC port is intended to provide an Ethernet connection to other network devices for example PC network HUB Two modes of operation are available 1 Bridge mode This mode allows the network traffics at the PC port to be bypassed to LAN port This means that the network device share the same network segment as the LAN port There is no IP address assigned to the PC port 2 Fixed IP This mode sets the PC port IP Addresss private and Subnet Mask manually This creates a new network segment for the network devices connected to the PC Port PC Port DHCP Server Enable 9 Disable To simplify network IP assignments enable the DHCP Server for the PC Port This allows network devices connected Port to obtain network IP and related information from the PC Port Please consult your network administrator for proper settings of the DHCP Server DHCP Server Enable Disable Starting Address 4220 Ending Address 14220 Static DNS optional Release 1 2 19 HT 84
37. pported Media 205 IPTOS Diffserv 5 Media Encryption For secure voice transmission RC4 ET263 Encryption methods are supported for the media channel Please make sure your service provider can support this encryption method before enabling this feature Media Encryption Non RC4 Encryption Media NAT Traversal 263 6 Symmetric RTP Enable the media channel to use symmetric RTP ports Some network environment demand the use of Symmetric RTP T Media NAT Traversal NAT Traversal can be set independently for Media packets This gives a more flexible setting for various network environment Three modes are supported STUN RFC 3489 Port forward DMZ and Relay Proxy Media NAT Traversal None None STUN REC 3482 Relay Proxy Release 1 2 30 HT 842R User Manual 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 Mode 1 Use UDP packets and encryption Mode 2 Use UDP packets and encryption use single UDP port Mode 3 Use TCP packets and encryption Use single TCP port The mode 2 and mode 3 are the passive and the port use is assigned by the RELAY SERVER Up to 4 backup Relay Servers are supported Once the designated Relay Server
38. s 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 and the rest must be digits from 0 to 9 and some commands require an operand To perform a Star Command dial the digit sequence on the phone set with DIMF dialing The corresponding tones are generated and transmitted to the device Once the code is dialed a timeout is initiated to wait for additional valid input If no input before the timeout expires the code is cancelled The default timeout during is 5 seconds 3 1 1 Star Commands for Phone Configuration The available star commands for Phone Configuration are listed in the tale below aes Function 0 ___ Generate a voice prompt on LAN Port IP assigned t reports Zero if the LAN IP is not assigned Generate a voice prompt on the PC Port IP assigned It reports Zero if the PC IP Address is not assigned or bridge mode 03 lt Operand gt Set LAN Port IP to lt Operand gt Operand XXX XXX XXX XXX 04 lt gt Set PC Port IP to lt gt lt Operand gt 09987456 Reset the IP Address for both LAN and PC ports LAN Port IP 192 168 0 1 Factory default is DHCP mode Release 1 2 3 HT 842R User Manual PC Port IP 19
39. sequence of digits If a specific digit has a limited range use the syntax A B where A and B are both digit to 9 and B is greater than A The length of the input number can be limited by using X to represent each unknown digit If this field is omitted it means any event action defines the action to be taken on the number received and it consists of minus plus and digits followed by digits means to remove the digits from the beginning of the number entered followed by digits means to add the digits in front of the number entered means or and the order of priority is from left to right Note For practical use it should not be possible to reach the maximum length of the Dial Plan string Examples 1 Dial Plan 010 010 means that the number dialed out will have the first digits 010 removed when a number with the first digits as 010 is entered a Number entered 01082121234 actual number dialed 82121234 b Number entered 82121234 actual number dialed 82121234 2 Dial Plan 1 00 means that the number dialed out will have the OO added in front of the number entered when a number with the first digit as 1 is entered a Number entered 1082121234 actual number dialed 00182121234 b Number entered 82121234 actual number dialed 82121234 Release 1 2 32 HT 842R User Manual
40. to Peer calls 4 Virtual Ringback This enables a ringback tone to be generated whenever call is made 5 DTMF Signaling This parameter sets the method of sending DTMF signals Inband measns that the signal is sent as an analog signal via the voice channel Outband means that the DTMF signal is sent as DIMF command via the data channel Both RFC2833 and SIP INFO methods are supported For RFC2833 a DIMF payload type is required and the default type is set to 101 Signaling Outband Qutband type 2833 RTP Payload 6 Signaling QoS This parameter sets the QoS mode for VoIP Signaling for better response time and more reliable VoIP Call signaling Both IP TOS and Diffserv modes are supported Please check with your network administrator or ISP for the correct setting Signaling 20 7 Signaling Encryption Five types of encryption methods are supported and these are used by various network equipment vendors in China to avoid blocking of SIP signaling traffics Please consult your SIP service provided to determine which encryption method is supported Signaling Encryption Non Signaling Traversal RIP Port Range acl a RC4 RC4 Encryption Key is required when it is enabled 0 Fast c VOS This encryption is developed by a network equipment vendor in Nanjing China Release 1 2 28 HT 842R User Manual d AVS T

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