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HT-912 User Manual

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2. lt Address rion Address HT 922 User Manual 4 3 3 Call Settings This page configures all related settings for VoIP Service Based on the two protocols 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 Advanced Settings Endpoint Mode Media Settings Config Single Contig Phone Humber Display Name H 323 ID Gatekeeper Address Enable vOS AVS Encryption Enable Authentication Dial Plan Fax Line 4 3 3 1 H 323 Phone The H 323 Phone selection for Endpoint Type refers to the protocol used Endpoint Type H 323 Fhane Endpoint Mode Gatekeeper Contig Mode Single Contig Phone Humber Display Mame H 323 ID Gatekeeper Address Enable VOS AVS Encryption Enable Authentication 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 20 HT 922 User Manual dialing the IP addresses or an alias Endpoint Type H 323 Phone a Endpoint Mode Direct Made Direct Gatekeeper Config Mode The device supports two modes Single Config Config by Line and Config by Group Singel Config allows only
3. Dial Pian Call Forward Type Not Forward CalFornwarddumbe ss Backup Server Enable Disable Fax Line HT 922 User Manual Config by Line Each FXS line has its own SIP registration and they are treated as independent lines a more reliable SIP Service Backup Server option is also available for each FXS line to insure 26 HT 922 User Manual Call Settings Line1 OLine2 Phone Number Phone Number 2 Display Name SIP Proxy SIP Registrar Server Register Expiry s Qutbound Proxy Home Domain Authentication ID Password Dial Plan Call Forward Type Not Forward Call Forward Humber Backup Server Q Enable Disable Fax gt The basic SIP settings are 1 Phone Number This parameter assigns the phone number used for SIP registration Phone Number 2 This parameter assigns the second phone number used for registration This acts as a second phone number for SIP calls Display Name This parameter optional specifies the Caller name and is transmitted as part of the caller ID 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 gt to the of the location For example If SIP Proxy yousippbx com the signaling port is the standard port 50
4. 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 Ring Cadence One 1200 Off 4000 ______ Off 5 Caller ID This defines the Caller ID signal to be sent from the FXS port It supports 35 HT 922 User Manual 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 Bellcore CID Signaling 1st Ring Duration 1000 CID 700 Delay Time After CID 1000 6 Hot Line When this feature is enabled the Hot Line Number defined will be dialed out automatically whenever the phone is off hook Hat Line1 Hot Line Enable Disable Hot Line Humber 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 is displayed when the saving is completed http 197 168 2 244 Configuration saved 36 HT 922 User Manual 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
5. There are four LEDs on the top case and they are described as follows DESCRIPTION RUN RED color 1 When the device is booting up the LED flashes at a rate of 100ms ON and 100ms OFF When the device has registered to the designated server the LED flashes ata rate of 1s ON and 1s OFF HT 922 User Manual 3 If the device fails to boot up the LED does not flash LAN Green color 1 on when there is a link with a network device 2 Flashes when there 15 data activities LINE 1 Red color 1 Flashes when there is a new message in the voice mail box LINE 2 Red color on when the phone connected is off hook 2 Turn off when the phone connected is on hook 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 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 DTMF 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 7 code is cancelled The default timeout during is 5 seconds 3 1 1 Star Commands
6. 00 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 3 Dial Plan 001 001 1751 means that the number dialed out will the first digits 001 changed to 1751 when a number with the first digits as 001 1 entered a Number entered 00182121234 actual number dialed 175282121234 b Number entered 82121234 actual number dialed 82121234 4 Dial Plan XXXX means that the input number is limited to 4 digit long and will be dialed out immediately when the fourth digit is entered 5 Dial Plan 13XXXXXXXXX 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 Plan 13 6 9 XXXXXXXX 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 7 8 0r 9 When this condition is matched the number dialed out will have a leading 0 added a Number entered 13901234567 actual number dialed 0139
7. Fast Start Enable Disable Register Mode Signaling Signaling 005 5 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 Q 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 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 registration message 23 HT 922 User Manual Register Multiple Times mode means that each number is registered in a separate registration message Register Mode Register Multiple Mur Register Multiple Register Multiple Times 6 DTMF Signaling This parameter sets the met
8. 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 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 32 Traversal Relay Proxy Address Port User Mame Password Relay Mode 1 2 3 4 Note Please contact your supplier for support ________ Encryptian Backup Relay Server Backup Relay Server Backup Relay Server Backup Relay Server provider see if this feature is available HT 922 User Manual For Service providers RELAY Proxy software is available at no charge For end user please contact your service 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 4 3 3 4 Dial Plan Audio Codec alaw ulaw 9729 q729a g729ab 97231 Dial Plan defines how the DEVICE processes a number 15 dialed This field is located in the Calling Setting Window and it is available for both H 323 and SIP modes The
9. PC Port IP Address 192 168 8 Subnet Mask o f DHCP Server 9 Enable Disable Starting Address 192 168 8 100 Ending Address 192 168 8 120 Static DNS optional Advanced HT 922 User Manual 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 for selection Network Configuration LAN Port PPPOE User DHCP Password PPPoE 802 1q VLAN 9 Enable Disable VLAN Id VLAN 005 O a Address em O Address 1 DHCP This mode should be selected If the network device functions as a DHCP host This allows the DEVICE to obtain all related network information settings from the DHCP host The factory default for DHCP mode is enabled and the 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 a Mask optional DNS optional 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 dial up is that the IP a
10. identifier containing an alphanumeric string Some gatekeepers may use this ID for authentication Gatekeeper Address This assigns the location of the Gatekeeper for VoIP Service 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 H 235 Auth If H 235 authentication is required enable this field and enter the H 235 ID and Password 1H 235 Auth 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 function properly or reliably 22 HT 922 User Manual Fax Line1 Fax Disable Advanced Settings More settings are available under the Advanced Settings tab These settings are common to all 323 configurations Depending on your network requirements please consult your network administrator for the correct configuration Advanced Settings RAS Port 0 931 Port H 245 Port
11. 60 If SIP Proxy yoursippbx com 15060 the signaling port is 15060 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 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 Registry Expiry s This specifies the expiry duration at the SIP Registrar after a 27 10 11 12 13 14 15 922 User Manual successful registration The range is 60 to 36400 seconds 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 15 available or not Home Domain This field enables the use of home domain name is SIP registration instead of IP address Authentication ID This field specifies the ID to be used for Authentication during a SIP registration Password This field specifies the password used for Authentication during a SIP registration 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 Forwa
12. 71234567 34 HT 922 User Manual b Number entered 13001234567 actual number dialed 13001234567 Please note that the above samples are simple and intended to show the meaning of various rules They may not have any practical meaning 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 Phone Settings PhoneBook Function FXS 48v Standby C Enable f Disable Billing Support C Enable Disable Ring Parameters Hat Line 1 Star Commands Star Commands for PhoneBook Function Default is 50 is programmable in this page The format 15 digits 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 1 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 48 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
13. Dial Plan is very flexible and can be configured for a wide range of dialing applications Dial Plan The basic syntax is lt event gt lt action gt lt event gt lt action gt where lt event gt defines the event to be matched A event consists of a sequence of digits lf a 33 HT 922 User Manual specific digit has a limited range use the syntax A B where A and B are both digit 0 to 9 and B is greater than A The length of the input number can be limited by using 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
14. M IP Phone Terminal b 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 c Time Server This parameter specifies the location of the network time server for obtaining the date and time information It accepts both domain name and IP address d Auto Provision This parameter enables or disables the Auto Provision procedures 15 HT 922 User Manual 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 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 Key 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 act
15. Route shows the IP address of the default gateway router that is used in the current network environment DNS Server 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 HT 922 User Manual 4 3 1 Preference This page configures the general settings in the device Language Time Zone Time server Auto Provision Key as Delimiter Auto dial Timeout Network Tones INFO Server China Phone Code Language r amp English i Key as Delimiter Enable Disable Time Zone Auto dial Timeout Time Server pool ntp org Network Tones China v Auto provision Q Enable 9 Disable INFO Server RemoteControl China Phone Code Q Enable 9 Disable Auto Reboot Reboot Time 04 00 a Language This 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 3C and Traditional Chinese 2 5 the language change is saved it does not take effect until the device is rebooted Language Z s fat Pr v To change the display language immediately you can select the language icon as shown below However this does not change the default language SEP
16. User Manual ANALOG TERMINAL ADAPTOR Model HT 922 Release 1 0 HT 922 User Manual Contents Overview 2 0 2 General qe o coo Uo o end qu Do OR M qM UE 3 E pem tob pi tap MEIN 3 1 5 Ww are SPCC I iod anie edat totu bul a tue ee 3 UU UU TT 1 1 9 ossa dara ud 4 21 Product View E 5 22 Connection ed uid 5 6 Basic aineas Star A 7 3 1 1 Star Commands for Phone Configuration 7 3 1 2 Star Commands for Phone Operation 8 TN 8 Do Phone Operation qim edu mm Id REESE RU E 9 331 Makea Pi mer ee ee ee ee 9 o 3 2 JANSWEl 9 3 3 3 Answer a Call Waiting Call not available in this model 9 3 04 Hold a Gall For only siesta eau ter ol aeta et ae teo is 10 3 9 9 bor SIP Only os aenea ses eod e Ea ade deris 10 3 3 6 Call Forward For SIP only
17. administrator level login ID admin is admin It is important to record the new password s If the admin password is lost a special star command is available to reset all system settings Please refer to section 3 1 1 for more information 38 HT 922 User Manual User Level Confirm Password fo 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 A 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 Tools Online Upgrade Change Password Reset Config Reboot 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 page is http xxx xxx xxx xxx en US gain html Enter this filed in a 39 HT 922 User Manual 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 dia
18. click on Save Changes to save all entries to memory The Name field is optional only the Number field is used for memory dialing IP Phone Terminal Humber Status Configurations Phone Book gqo Ji Save Changes o Tools el H JlLl 1 4 5 Tools The Tools section is intended to offer the following functions Online Upgrade Change Password Reset Config and Reboot 37 HT 922 User Manual 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 Online Upgrade Current Version 38H5 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
19. conds f1 is the tone 1 300 3000 2 12 is the tone 2 300 3000 2 f3 is the tone 3 300 3000 Hz 14 is the tone 34 300 3000 2 p1 is the attenuation index for tone 1 0 31 p2 is the attenuation index for tone 2 0 31 p3 is the attenuation index for tone 3 0 31 p4 is the attenuation index for tone 4 0 31 0 3dB 1dB increments 023GB 1dB increments 0 3dB 1dB increments 0 3dB 1dB increments m gt gt gt gt Two examples of network tone definition are shown below 1 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 New Zealand Busy tone 400Hz with cadence of 500ms 500ms off repeat is defined as 1 0 500 500 0 0 0 0 400 0 0 0 10 0 0 0 i INFO SERVER This parameter enables the HT 922 to forward the caller ID of all incoming calls to the server IP specified j China Phone Code This parameter enables the HT 922 to use the China area codes programmed and dial the phone number out automatically once the preset conditions are matched There is no need to dial the key when this mode is enabled k Auto Reboot This parameter programs the HT 922 to perform a cold boot daily at the time specified 4 3 2 Network Configuration This page configures the network interface for LAN Port and PC Port Network Configuration LAN Port Specify ONS server 802 19 VLAN Enable 9 Disable
20. ddress obtained for the LAN port is normally a public IP HT 922 User Manual Network Configuration LAN Port User More advanced parameters for 802 1q VLAN and MAC settings are 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 Fixed IP This mode sets the PC port IP Addresss private IP and Subnet Mask manually This creates a new network segment for the network devices connected to the PC Port PC Port Static IP IP Address 192 168 8 Subnet Mask DHCP Server Enable Disable Starting Address 192 168 8 100 Ending Address 192 168 8 120 Static DNS optional The above figure shows the factory default settings for the PC port To simplify network IP assignments the DHCP Server for the PC Port is enabled 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 Both MAC Address and IP Broadcast Address can be set in the Advanced tab
21. determine if this mode is supported 24 HT 922 User Manual signaling NAT lone Traversal Nat Citron RTP Port range Port forward DMZ Relay Proxy 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 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 In SIP Phone mode it supports two registration modes Single Server and Config by Line Single Server mode only one SIP registration is used for both FXS lines The HT 922 will perform line hunting automatically when a call is made or a call comes in 23 Call Settings Endpoint Type SIP Phone Config Mode Single Server Mode phonenumber _ PhoneNumber2 Display Name SlPRegistrarsever __________ Register Expiry s Home Domain _ AuthenticationID mswa
22. e to network conditions in addition to the sequential mode 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 supported Media Qos 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 31 HT 922 User Manual Media Encryption None bw Encryption Media Traversal ETZ53 6 Symmetric RTP Enable the media channel to use symmetric RTP ports Some network environment demand the use of Symmetric RTP 7 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 RFC 3489 Fort forward OMZ Relay Proxy 8 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 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
23. e PC IP address when 1 PC and the PC port HT 922 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 192 168 2 134 in Web Browser as shown below IE Firefox etc e MSN Hong Kong Homepage Windows Internet Explorer T Iv 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 HT 922 User Manual Connect to 192 168 2 134 The server 192 168 2 134 at Please Login requires username and password Warning This server is requesting that Your username and password be sent in an insecure manner basic authentication without a secure connection _ Remember my password Cancel 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 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 isa Star 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 Once the login is successful the W
24. eb Browser brings up the Status page as shown below I N sy D IP Phone Terminal LIEU WV status fe SMe AI Phone Information Network Information Configurations Serial Number HTS1 2080500028 LAN Port 182 158 2 134 Firmware Version amp 38H5 3 15 LAN MAC Dn0 11 BE 02 0F CB Phone Book Hardware Model 1 fs PC Port In Bridge Made Toole minal 1 LOGOUT PPPoE Disabled Default Route 192 158 2 1 DNS Server 202 130 97 65 13 HT 922 User Manual 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 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 If the device registers to the designated server s successfully it displays the status LOGIN Otherwise it displays LOGOUT 4 2 2 Network Information LAN Port This field shows IP address assigned to the LAN port PC Port This field shows IP address assigned to the PC port PPPoE This field shows the dial up status when PPPOE is enabled for ADSL login Default Route The Default
25. empty the entry is considered as empty NO Name Number 01 jonsoon 02 03 ony 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 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 for more information 11 HT 922 User Manual 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 Us
26. er Manual DSP for voice codec and voice processing Two 10 100M Based Ethernet ports for WAN LAN connections LED status indicators Two FXS ports 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 922 FXS Gateway b One AC DC Adapter DC24V 300mA c One Ethernet cable 3 Meter long HT 922 User Manual 2 Installation The device is designed for easy installation and can be installed in various network environments 2 1 Product View and Interface 1 LINE1 Connect this port to a traditional telephone set or a PBX trunk line via a standard telephone cable 2 LINE2 Connect this port to a traditional telephone set or a PBX trunk line via a standard telephone cable 3 LAN Connect this port to a network device Broadband modem router with internet access via a RJ 45 Ethernet cable 4 PC Connect this port to an Ethernet Hub or PC a RJ 45 Ethernet cable 5 POWER Connect this port to a 224VDC 30mA AC DC Adapter provided 2 2 Connection Diagram Install the device as shown in the connection diagram below HT 922 User Manual E ADSL Cable Model Ethemet Phone 2 3 Indicators
27. for Phone Configuration The available star commands for Phone Configuration are listed in the tale below Star Command Function 0 0 0 0 Generate a voice prompt on LAN Port IP assigned t reports Zero if the LAN IP is not assigned HT 922 User Manual 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 lt Operand gt 04 lt Operand gt Set PC Port IP to lt Operand gt lt Operand gt 20 Send a remote support request 09987456 Reset the IP Address for both LAN and PC ports LAN Port IP 192 168 0 1 Factory default is DHCP mode PC Port IP 192 168 5 1 Facotry default is bridge mode 11983185922 Reset all system parameters back to factory defaults 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 Hold the current call Release the Hold call 41 Call Transfer to another VoIP Number 50 Phone book function key 3 2 Setup The device supports two major configuration
28. gnaling 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 sent to the local firewall router in order to keep the ports opened for VoIP service 3 P2P his enables Peer to Peer calls 4 Virtual Ringback This enables a ringback tone to be generated whenever a call is made 5 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 Outband means that the DTMF signal is sent as DTMF command via the data channel Both RFC2833 and SIP INFO methods are supported For RFC2833 a DTMF payload type is required and the default type is set to 101 DTMF Signaling Outband Outband type 2833 RTP Payload Type 101 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 29 HT 922 User Manual Signaling QoS 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 M
29. hod of sending DIMF signals Inband measns that the DIMF signal is sent as an analog signal via the voice channel Outband means that the signal is sent as DIMF command via the data channel and is commonly known as RFC2833 n Outband mode DTMF payload type is required and the default type is set to 101 DTMF Signaling Outhand Qutband 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 QoS 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 methods supported are NAT 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
30. hone 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 HT 922 User Manual 3 3 4 Hold a Call For SIP only Dial the Star Command the default is 42 to put the active call on hold release the 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 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 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 dia
31. l 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 A 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 If 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 Answer This feature requires the support from the VoIP service provider and is enabled in the built in web server HT 922 User Manual 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 a number or name If this field is
32. ling may not be detected properly You can always click on Reset to resume to system default level Line 1 Line 1 Output Gain Line 1 Input Gain dB Line 2 Line 2 Output Gain Line 2 Input Gain Ode 40
33. 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 TFIP server This configuration file contains all the necessary parameters to set up the device for VoIP Services This HT 922 User Manual 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 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 Web Browser Please refer to Section 4 for more detailed information 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 Just dial a valid VoIP number and then to make a call If the 7 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 p
34. n A VoIP FXS Gateway Analog Telephone Adapter ATA is a telephone extension to the IP network It offers a traditional telephone line PSTN interface to an analog telephone PBX line extension or a fax machine Its WAN port interface allows access to the IP network in order to offer voice and fax services It is a great way for turning a traditional PBX to access the low cost VoIP services and for deploying VoIP service by a ISP An additional Ethernet allows broadband connection by the existing PC without buying additional network equipment The HT 922 referred as the device offers the features mentioned above and is an ideal low cost solution for SME and SOHO IP telephony application 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 HMAC SHA1 ITU T G711 Alaw ULaw G 729A G 729AB G 723 1 Voice Codec HFC1889 Real Time Data Transmission Proprietary Firewall Pass Through Technology SIP V2 0 Standard Simple 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 Relay RFC 2833 and SIP Info 1 3 Hardware Specification ARMSO9E Processor for high performance HT 922 Us
35. on Signaling NAT Travers al RIP Port Range a RC4 RC4 Encryption Key is required when it is enabled b Fast c VOS This encryption is developed by a network equipment vendor in Nanjing China d This encryption is developed by a network equipment vendor in Shanghai China e 263 This encryption is supported by major network equipment vendors 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 NAT Traversal mode for VoIP signaling 2 methods supported STUN RFC3489 and Relay Proxy ASTUN Server is required for STUN RFC3489 Signaling Traversal STUN REC 3489 Relay Proxy 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 30 HT 922 User Manual Note For Service providers RELAY Pr
36. one phone number and gatekeeper configuration Config Mode Single Config Phone Number f Display Name fs 1 325 1D gatekeeper Address f Config by Line allows each FXS line to have its own configuration of the following parameters phone number H 323 ID Gatekeeper Address Encryption method for China Market see Config by Group H 235 Auth Dial Plan see section 4 3 3 4 and Fax Line Config Mode Config by Line Line 1 O Line 2 phonenumber sd 43230 Gatekeeper Address E Enable VOS AVS Encryption C H 235 Auth Dial Plan Fax gt Config by Group allows up to 4 groups of configuration for phone number H 323 ID and gatekeeper Since there is only one FXS port all four groups are sharing the same FXS port 21 HT 922 User Manual Contig Mode Config by Group Group 1 C Group 2 C Group 3 C Group 4 Phone Humber Gatekeeper Address Enable 5 5 Encryption 4 235 Auth In order to activate the FXS line to be used the following parameter must be checked for each group Activated Lines in Group 1 Fii 112 Fax gt Phone Number This parameter assigns the phone number used for registration in Gatekeeper Mode This is used as an alias in Direct Mode Display Name This parameter optional specifies the Caller name and is transmitted as part of the caller ID H 323 ID This parameter is specified in the H 323 protocol It is an
37. oxy 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 VoIP call is established the Media channel is used for voice transmission The settings listed below configure the performance and operation of the Media channel Media Setings RIP Port Range 6384 B27 68 PacketLength ms 0 Jitter Buffer Delay ms Media QoS Media Encryption Symmetric RTP Media NAT Traversal None 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 15 16384 32768 Specify the port range depending on your network environment if needed Packet length ms This specify the length of a voice packet The default packet length is 20 ms 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 respons
38. rd Call is forwarded when the line is in use engaged d No Answer Forward 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 Call Forward Type Mot Forward j Unconditional Forward Call Forward Number Backup Server Busy Forward Answer Forward Busy or Ma Answer Forw Forward Number This defines the number to be used for Call Forward 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 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 function properly or reliably Fax Line1 Fax Disable Advanced Settings More settings are available under the Advanced Settings tab Depending on your 28 HT 922 User Manual network requirements please consult your network administrator for the correct configuration Advanced Settingss Signaling Port 5060 NAT Keep alive Enable Disable 2 C Enable Disable Virtual Ringback C Enable Disable Advanced Timing DTMF Signaling Outband type RTP Payload Type Signaling 005 Signaling Encryption Traversal 1 Si
39. ually 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 Auto Dial Timeout This parameter sets the maximum delay for sending out a call request after the last number dialed The range is 3 to 15 seconds and the default is 5 seconds 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 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 Customized Dial Tone o f Ring Back Tone Indication Tone Doo Each network tone is defined as nc rpt clon c2on c2off c3on c3off f1 12 13 f4 p1 p2 p3 p4 where nc is the number of cadences rpt is the repeat counter 0 infinite 1 n repeat 1 n times c1on is the cadence one on duration in milliseconds HT 922 User Manual is cadence one off duration in milliseconds c2on is the cadence two on duration in milliseconds c2off is the cadence two off duration in milliseconds c3on is the cadence three on duration in milliseconds c3off is the cadence three off duration in millise

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