FXO User Manual
Contents
1. 82121234 b Number entered 82121234 actual number dialed 82121234 2 Dial Plan 1 00 means that the number dialed out will have the 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 3 digits O01 changed to 1751 when a number with the first digits as O01 is 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 S Dial Plan 1 3XXXXXXXXX 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 be2. inserted in this number a dialing pause of 500 ms is inserted when dialing the number to the PSTN line 3 Forward Password VoIP to PSTN When it is set the forward password from VoIP to PSTN feature is enabled When a VoIP caller is trying to access the PSTN line an indication tone is first generated to prompt the caller to dial the password Once the correct password is dialed the PSTN line is connected and the caller can then hears the PSTN dial tone If Forward Number VoIP to PSTN is set the number will then be dialed out automatically 4 Dial Plan VoIP to PSTN Please refers section 3 3 5 for more information 5 Forward to VoIP This enabled incoming PSTN calls to be routed to the VoIP network When it is disabled no incoming PSTN calls are accepted When it is enabled incoming PSTN calls are first answered and then processed according to the 3 parameters below 6 Forward Number PSTN to VoIP When a VoIP number is assigned here incoming PSTN calls are forwarded to VolP number automatically 7 Forward Password PSTN to VoIP When it is set the forward password from PSTN to VoIP feature is enabled When an incoming PSTN call is answered an indication tone is generated to prompt the user to enter the forward password entered in this field Once the Release 1 0 25 Web Configuration HT 342 User Manual correct password is entered a VoIP dial tone is generated to allow making a VoIP call If the Forward Numbe
3. 6 7 8 or 9 When this condition is matched the number dialed out will have a leading 0 added a Number entered 13901234567 actual number dialed 013971234567 Release 1 0 29 Web Configuration HT 342 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 A combination of these rules joined with the symbol can be realized for a much more complicated dialing application 3 6 Gain Settings The gain setting must be cautious to using lt was a hidden web page If you want adjust the VoIP Lines s volume Please rewrite the URL address to htto xxx xxx xxx xxx default en_US gain html and enter The web browser will show up a GAIN SETTINGS screen You can adjust the volume of the two PSTN Lines to different values The range you can adjust is from 5 to 5 E IP Phone Gateway HERE ABE BA TAM ABH Qa O HAO Ser ee O 2 0 JEBAD Ss HE DO http 192 168 8 1 default en_ US gain html AL S hone G SY de IP Phone Terminal Gain Settings Line 1 Line 1 Output Gain Line 1 Input Gain 2 wt Line 2 Line 2 Output Gain Line 2 Input Gain 2 hal Note A too low or too high input gain will make DTMF detector insensitive Release 1 0 30
4. RFC 3489 Port forward DMZ and Relay Proxy Media NAT Traversal None None STUN RFC 3489 Port forward DMz Relay Proxy 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 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 Release 1 0 23 Web Configuration Media NAT Traversal Address User Name Password L Encryption Relay Mode Backup Relay Server o _ Relay Server II Relay Server E Relay Server E A HT 342 User Manual Note For Service providers RELAY Proxy software is available at no charge Please contact your supplier for support if this feature is available For end user please contact your service provider to see 9 Audio Codec Preference The table below list the voice codec priorities in de
5. 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 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 None iw Signaling NAT Traversal RIP Port Range acl ntl riartha ich 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 AVS This encryption is developed by a network equipment vendor in Shanghai China e ET263 This encryption is supported by major network equipment vendors in China 8 Signaling NAT Traversal NAT Traversal is an algorithm designed to solve a common problem in TCP IP 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 2 methods supported are STUN RFC3489 and Relay Proxy A STUN Server is required for STUN RFC3489 Release 1 0 21 Web Configuration HT 342 User Manual Signaling NAT none ne Traversal STUN REC 3489 Relay Proxy Relay Proxy mode is a
6. a number when it 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 Dial Pian Release 1 0 28 Web Configuration HT 342 User Manual 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 Ifa 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 X to represent each unknown digit If this field is omitted it means any event lt action gt 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 di Dial Plan 010 010 means that the number dialed out will have the first 3 digits 010 removed when a number with the first digits as O10 is entered a Number entered 01082121234 actual number dialed
7. and data activities occurs PC This LED shows the PC port status It flashes when link and data activities occurs FX01 This LED shows the line status of the corresponding FXO port The LED lights up when the corresponding FXO line is OFF hook state In Use FX02 This LED shows the line status of the corresponding FXO port The LED lights up when the corresponding FXO line is OFF hook state In Use FX03 This LED shows the line status of the corresponding FXO port The LED lights up when the corresponding FXO line is OFF hook state In Use FX04 This LED shows the line status of the corresponding FXO port The LED lights up when the corresponding FXO line is OFF hook state In Use The HT 342 has four FXO ports two Ethernet ports LAN and PC Power Input and a Reset switch They are all located at the back panel of the HT 342 as shown and described below Release 1 0 3 Installation HT 342 User Manual IS HT 342 Rear View mb FXO 1 This is a FXO port and is to be connected to a PSTN line 2 FXO 2 This is a FXO port and is to be connected to a PSTN line 3 FXO 3 This is a FXO port and is to be connected to a PSTN line 4 FXO 4 This is a FXO port and is to be connected to a PSTN line 5 LAN This Ethernet port is intended for network access It can be connected to a network switch xDSL modem or other network access equipment 6 PC This Ethernet port is intended for connecting to
8. bandwidth change and status messages between two H 323 endpoints If not specified the port address is assigned automatically 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 It should be specified as a port range If not specified the port address is assigned automatically 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 Register Mode Two registration modes are support Register Multiple Numbers mode means that multiple numbers are registered in a single registration message Release 1 0 16 Web Configuration HT 342 User Manual Register Multiple Times mode means that each number is registered in a separate registration message Register Mode Register Multiple Mur Register Multiple Numbe Register Multiple Times 6 DTMF Signaling This parameter sets the method of sending DTMF signals Inband measns that the DTMF signal is sent as an analog signal via the voice channel Outband means that the DTMF signal is sent as DIMF comm
9. 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 3 3 3 3 Media Setting 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 6304 2768 PacketLengthims O Jitter Buffer pelayo Media QoS Media Encryption None iw MW Symmetric RTP Media NAT Traversal None 1 RTP Port range Audio stream is transmitted via Real Time Protocol RTP and at least 4 ports are used per voice channel The default port range is 16384 32768 Specify the port range depending on your network environment if needed 2 Packet length ms This specify the length of a voice packet The default packet length is 20 ms 3 Jitter Buffer Mode Three jitter modes are available The Fi
10. CMP TFTP Client Release 1 0 1 HT 342 User Manual 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 1 3 Hardware Features ARMGQE 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 Features 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 Caller ID PSTN to VoIP Multiple Language Support 1 5 List of Package The following items are included in the package 1 One HT 342 main unit 2 One DC12V 500mA power adaptor 3 One Ethernet cable 8M Release 1 0 2 Installation HT 342 User Manual 2 2 1 Installation Appearance HT 342 Front View There are eight LEDs on the front panel to display the current status of the HT 342 Power This LED lights up when the power is applied 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 lt flashes when link
11. HT 342 User Manual VoIP FXO Gateway User Manual Model HT 342 Release 1 0 Table of Content HT 342 User Manual T INTRODUCTION sara 1 LL WON ERWIEW cata ias 1 12 PROTOCOL Sidonia 1 HL Se PHAR D WARE FEATURES ra oi 2 1 4 SOFTWARE FEATURES a acia 2 5 SPORE PACKAGE cdt ola tacto 2 2 INSTALLATION vara Ea 3 ZA APPEARANC Eoi ie ie 3 ze GONNECTION sao 4 3 WEB CONFIGURATION as 6 SL ACCESS THE BUILT IN VV EB SERVER aiii tetas 6 A saialauah oe a agian E gee Os unde ss Taha eead A es rea sas lon 8 FZ L FOE TOMI AOU a AL AAA AA alas add SAAD doth ath T 8 See NOLWO KIMONO E A OE Pantene asias 8 5 5 CONFIGURATION esans T ias 8 PO LANA 9 Oe INCIWOTK CORTIQUIATION oia i AET E ee 11 Fo GANSOS valia tilde E EN O EA 13 ol POLS OMG aiena A A Aa 13 DO SINO O idiota 18 DIOS MEDIOS CUINA o ciated 22 i Ore CANONE tie 25 i000 DAVE CONGU AVO ii A A a A A 26 390 00 DISCO CM GAG Si A tia 26 MODE a bl cunts pal acest 26 O AB lote AE PAM O Po AE E S E E ener ee er rer Meee terete 27 S42 Ghand Ras WO iio n 27 343 OSCE Cono UON ri ts 28 2AA A A Y 28 SS Ia an Na 28 3 30 GAIN SETIINGS dao sets 30 Release 1 0 i Introduction HT 342 User Manual 1 Introduction 1 1 Overview A VolP FXO Foreign Exchange Office Gateway bridges the VolP and PSTN networks by enabling both voice and fax communications It offers an FXO interface to a traditional telephone line PSTN and an WAN Ethernet port interface to the IP Network A connect
12. a PC or other equipment that requires network access 7 POWER DC12V 500mA This power jack is connected to the power adapter provided 8 Reset This is the reset switch for HT 342 2 2 Connection Please follow the connection diagram shown below to install the HT 342 FXO Gateway EET mm p SONS A e Ma AC DC Power Adaptor Use the one provided or equivalent AP PSTH Lines Local Network Connection PC Other Network Device External Network Connection xDSL Cable modem Router Gateway etc Release 1 0 4 Installation HT 342 User Manual 1 Connect a PSTN line to each FXO port FXO1 and FXO2 2 Connect an Ethernet cable or the cable supplied to the LAN port and the other end of the cable is connect to an equipment with network access 3 Connect the PC port to a PC or other equipment Optional 4 Connect analog phones to both Line1 and Line2 phone jacks 5 Insert the adapter plug provided to the Power jack and then plug the adaptor to an AC wall plug PLEASE ONLY USE THE ADAPTER PROVIDED OR AN ADAPTOR WITH THE SAME SPECIFICATION Release 1 0 5 Web Configuration HT 342 User Manual 3 Web Configuration The HT 342 can be configured via its built in Web Server HTML To access this web server either the LAN IP or the PC IP must be known The LAN port is pre configured to DHCP client mode and the PC port IP is set to 192 168 8 1 3 1 Access the Built in Web Server The bui
13. 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 Release 1 0 9 Web Configuration HT 342 User Manual executed at boot up time and is repeated at a duration specified in the parameter Provision Interval Auto provision 2 Enable Disable 5 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 Dial Tone Ring Back Tone Indication Tone Each network tone is defined as nc rpt clon cloff c2on c2off c3on c3off f1 f2 f3 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 c1off 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 milliseconds f1 is th
14. and via the data channel and is commonly known as RFC2833 In Outband mode a DTMF payload type is required and the default type is set to 101 DTMF Signaling Outhband w Qutband 7 Signaling QoS This parameter sets the QoS mode for VoIP Signaling for better response time and more reliable VolP 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 The 3 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 determine if this mode is supported Signaling NAT Non a Traversal Nat Citron RTF Port range Fort forward DMZ Relay Proxy Release 1 0 17 Web Configuratio
15. e sent in an insecure manner basic authentication without a secure connection 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 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 Web Browser brings up the Status page as shown below Phone Information Network Information serial Number HT32207070008 LAN Port 192 166 2275 Firmware Version 034HS 3 01 1 LAN MAC 00 11 BE 01 BAT 4 Hardware Model afso PC Port 192 168 8 1 Line Register 1 LOGOUT PPPoE Disabled Status Line Register 2 LOGOUT Default Route 192168 2 254 Status DNS Server 2027 96 134 133 Release 1 0 7 Web Configuration HT 342 User Manual 3 2 Status The Status page shows the current status hardware and software information of the HT 342 3 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 tec
16. e tone 1 300 3000 Hz f2 is the tone 2 300 3000 Hz f3 is the tone 3 300 3000 Hz f4 is the tone 34 300 3000 Hz 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 0 3dB 1dB increments 0 3dB 1dB increments 0 3dB 1dB increments ee ee m ee Na a a NSN Two network tone definition samples are shown below 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 repeat is Release 1 0 10 Web Configuration HT 342 User Manual defined as 1 0 500 500 0 0 0 0 400 0 0 0 10 0 0 0 6 PSTN Has Line Reversal When this parameter is enabled the HT 342 enables the fast start channel in H 323 or the Early Media channel in SIP to monitor when the called party answers the call This feature is commonly used to achieve more accurate billings for calls to the PSTN line 3 3 2 Network Configuration This page configures the network interface for LAN Port and PC Port Network Configuration LAN Port PPPoE wt PC Port static IP e 802 1q VLAN Enable Disable Advance gt gt 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 m
17. eral 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 lt 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 Release 1 0 27 Web Configuration HT 342 User Manual User Level NewPasswort 7 Confirm Password 2 Change Administration Level new Password Confirm Password Change 3 4 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 3 4 4 Reboot Click on the Reboot tab to reboot the device 3 5 Dial Plan Dial Plan defines how the DEVICE processes
18. eration 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 IP and Subnet Mask manually This creates a new network segment for the network devices connected to the PC Port PC Port Static IP DHCP Server O Enable Y 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 Release 1 0 12 Web Configuration HT 342 User Manual DHCP Server Enable Disable Starting Address Po Ending Address Po Static DNS optional 3 3 3 Call Settings This page configures all related settings for VoIP Service Based on the two protocols H 323 and SIP supported the operation of DEVICE is divided as two Endpoint Types H 323 Phone and SIP Phone Some of the parameters are unique to the Endpoint Type and are described separately below Call Settings Endpoint Type H 323 Phone dll Advanced Settings Endpoint Mode Config Mode Single Contig Phone Number fo Display Name Po batekeeper Address fo DJ Enable VOS AVS signaling Encryption Enable Authentication Fax Line g
19. hnical support and warranty repair The product label at the bottom also contains this information Firmware Version This field identifies the current Firmware Version installed Hardware Model This field identifies the hardware model and version 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 3 2 2 Network Information LAN Port This field shows IP address assigned to the LAN port LAN MAC This field shows the MAC 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 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 3 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 Release 1 0 8 Web Configuration HT 342 User Manual 3 3 1 Preference This page configures the general settings in the device Language Time Zone Time server Auto Provision Key as Delimiter Auto dial Timeo
20. ion can be initiated either from a VoIP user to a PSTN user or from a PSTN user to a VoIP user It is a key component for building a hybrid communication system with connections to both VoIP and PSTN networks The key advantage is to make use of the IP network and local PSTN networks to save on international long distance phone expenses The HT 342 is designed as a compact high performance and low cost FXO Gateway It features superb audio quality reliable FXO detection rich functionalities and high level integration The FXO detection is optimized to avoid the hold up of the PSTN line when the other party is disconnected This has been one of the key issues in the design of FXO gateway The incoming PSTN Caller 1D is also transmitted to the VolP user for more user friendly operation The HT 342 is a full featured FXO gateway with a second Ethernet port for connection with another network device Itis an ideal solution for VolP to PSTN termination in both SME and SOHO environment 1 2 Protocols TCP IP V4 IP V6 auto adapt ITU T H 323 V4 Standard H 225 V4 Standard H 245 V7 Standard H 235 Standard MD5 HMAC SHA1 ITU T G 711 Alaw ULaw G 729A G 729AB and G 723 1 Voice Codec RFC1889 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 I
21. lt 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 Via the LAN port 1 APC connected to the same network segment as the LAN port is available Please note that the PC port is in the same network segment as the LAN port if it is set to bridge mode 2 The LAN port IP address must be known If the LAN port has a public IP to the internet any PC with internet access can access its built in web server Via the PC Port 1 The PC port is pre configured to the fixed IP address 192 168 8 1 and the DHCP server is not enabled 2 Inthis case the PC connected to the same PC port network segment the PC IP address must be configured to an IP address that is in the same network segment Please consult the User Manual of your computer OS for configuring the PC IP address To access the built in Web Server type the correct IP address of the HT 342 in a Web Browser IE Firefox etc as shown below E MSN Hong Kong Homepage Windows Internet Explorer TSE http 192 168 2 134 Once the device responds to the HTTP request the Web Browser will prompt for a login wndow as shown below Release 1 0 6 Web Configuration HT 342 User Manual Connect to 192 168 2 134 The server 192 168 2 134 at Please Login requires a username and password Warning This server is requesting that your username and password b
22. n HT 342 User Manual 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 3 3 3 2 SIP Phone The SIP Phone selection for Endpoint Type refers to the SIP protocol used 1 Config Mode The device supports two modes Single Server Mode and Config by Line Config Mode single Server Mode Single Server Mode Contig by Line Single Server mode only one SIP registration is used for both FXO lines The HT 342 performs line hunting automatically when a call is made to the PSTN lines Config Mode single Server Mode Phone Number Phone Number 2 Display Name SIP Registrar Server Register Expiry s Outbound Proxy Home Domain AuthenticationID Dial Plan Call Forward Type Call Forward Number Backup Server O Enable 6 Disable Fax Line gt Release 1 0 18 Web Configurati
23. odes are available for selection Network Configuration LAN Port PPPOE m User name Password PPPoE 802 1q VLAN Enable Disable VLAN id VLAN Qos Advance a hd Address sra IO 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 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 Release 1 0 11 Web Configuration HT 342 User Manual Network Configuration LAN Port Static IP es tc O Mask optional DN S 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 address obtained for the LAN port is normally a public IP Network Configuration LAN Port User name 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 op
24. on HT 342 User Manual Config by Line each FXO line has its own SIP registration and they are treated as independent lines A Backup Server option is also available for each FXO line to insure a more reliable SIP Service E A Call Settings Endpoint Type SIP Phone we Config Mode Config by Line e Line 1 ObLine 2 Phone Number Phone Number 2 Display Name SIP Proxy SIP Registrar Server Register Expiry s5 Outbound Proxy Home Domain Authentication ID Password Dial Plan Call Forward Type Mot Forward AU Call Forward Number Backup Server O Enable Disable Fax Line Phone Number This sets the phone number to be used in SIP registration and calls Phone Number 2 A second number can be assigned for second SIP registration Display Name This assigns the name to be used in the Caller ID name delivery SIP Proxy This sets the SIP proxy in either IP Address or domain name format The standard port used for SIP signaling is 5060 To specify a different port add colon after the SIP Proxy address and then the desired port number e g sip at338 com 5070 SIP Registrar Server This sets the SIP Registration Server either IP Address or domain name format If this is blank the SIP Proxy will be used for registration The standard port used for SIP signaling is 5060 To specify a different port add colon after the SIP Proxy address and then the desired port number e g sip at338 com 5070 Regi
25. r Backup Home Domain Il 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 m Advanced Setings Signaling Port NAT Keep alive Enable Disable P2P O Enable Disable Virtual Ringback O Enable Disable Advanced Timindg DTMF Signaling QOutband DTMF type RTP Payload Type Signaling Qos Signaling Encryption tad Traversal Signaling Port This Port is used to convey signaling message with the SIP Proxy The standard port number is 5060 NAT Keep alive When enabled a dummy packet sent to the local firewall router in order to keep the ports opened for VoIP service P2P This enables Peer to Peer calls Virtual Ringback This enables a ringback tone to be generated whenever a call is made DTMF Signaling This parameter sets the method of sending DIMF signals Inband measns that the DTMF signal is sent as an analog signal via the voice channel Outband Release 1 0 20 Web Configuration HT 342 User Manual means that the DIMF signal is sent as DIMF 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 DTMF type RFC 2833 RTP Payload Type 101 6 Signaling QoS This parameter sets the QoS mode for VoIP
26. r PSTN to VoIP is set this number is dialing out automatically 8 Dial Plan PSTN to VoIP Please refer section 3 3 5 for more information 9 VoIP Auto Answer Time s This sets the waiting time before answering an incoming PSTN call 3 3 3 5 Save Configuration Once a change is confirm users should click on the Save Configuration button in the Configuration page Otherwise your configuration will not take effect After user click the Save configuration the screen will be D gt Configuration saved 3 3 3 6 Discard Changes Status Configurations Preferences Network Call Settings save Changes 3 4 Tools The Tools section is intended to offer the following functions Online Upgrade Change Password Reset Config and Reboot Release 1 0 26 Web Configuration HT 342 User Manual Tools Online Upgrade Change Password Reset Config Reboot 3 4 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 ASGHS 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 3 4 2 Change Password The device supports two login levels to the built in webpage The User level is intended for gen
27. r for VoIP Service 8 VOS AVS Signaling 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 WOS Encryption supports two modes Signaling Encryption and Signaling and Media Encryption Please consult your services provider for more information 9 Authentication If H 235 authentication is required enable this field and enter the H 235 ID and Password Release 1 0 15 Web Configuration HT 342 User Manual 10 11 12 1H 235 Auth H 235 ID Password Dial Plan This defines the dialing rule when making PSTN calls Activate Lines in Group X This section specifies the FXO lines to be used in the Group Fax Line Enable the support of FAX function for the line selected Advanced Settings More settings are available under the Advanced Settings tab These settings 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 E Fast Start Enable Disable Register Mode DTMF Signaling Signaling QoS Traversal Media Settings RAS Port This Port is used to convey the registration admissions
28. scending 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 0 Audio Codec Preference Malan wl lev a Y 9729 g729a a729ab 97231 Media Settings lt RTP Port Range PacketLength ms Jitter Buffer Delay ms 60 Media QoS None Media Encryption Mone i L Symmetric RTP Media NAT Traversal None 24 Web Configuration HT 342 User Manual 3 3 3 4 Call divert This page sets the call routing to and from the PSTN lines Each line has its own settings Select the line desired and then program the parameters below Call Divert 6 Linet Line Forward to PSTN Enable Disable eae Cis VoIP To PSTN cen cae Doo O VoIP To PSTN Dial Plan VolP to Po PSTN Forward to VoIP Enable Disable Forward Number PSTN To VolP Forward Password PSTN To VolP fF E Dial Plan P STN to O VoIP VolP Auto Answer 1 Forward to PSTN This enables or disables calls to be forwarded to the PSTN line When it is set to disabled no VoIP calls are routed to the PSTN line 1 When it is enabled VoIP calls are forwarded to the PSTN line according the following 3 parameters 2 Forward Number VoIP to PSTN When a PSTN number is assigned VoIP calls are forwarded to PSTN via the corresponding line automatically When a comma is
29. ster Expiry s This sets the registration period Outbound Proxy An Outbound proxy is mostly used in presence of a firewall NAT to handle the signaling and media traffic across the firewall If the SIP proxy can handle NAT or has a built in outbound proxy this field can be set to the same address as the SIP proxy or left blank In some cases the outbound proxy referred as Outbound Session Controller is placed alongside the firewall and is the only way to let SIP traffic pass from the internal network to the Internet In these cases the Outbound Proxy must be set properly in order to ensure normal Release 1 0 19 Web 10 11 12 13 14 Configuration HT 342 User Manual SIP operation Home Domain This sets the domain name for providing service to a SIP user Typically this is the domain present in the URI in the address of record of a registration Authentication ID This is used for authentication during registration Very often the Phone Number is used as Authentication ID Password This field specifies the password for authentication Call Forward Type This sets the type of Call Forwarding modes Call Forward Number This sets the number to be used in Call Forwarding Backup Server When system support backup server fill in the messages of the backup server When the first server is failed HT 342 will try the backup server Backup Server Enable Disable Backup SIP Proxy Backup SIP Registra
30. t gt Media settings gt gt 3 3 3 1 H 323 Phone The H 323 Phone selection for Endpoint Type refers to the protocol used The basic H 323 settings are 1 Endpoint Mode Gatekeeper Mode supports VoIP calling via a call server Server Registration is required Direct Mode supports making a VoIP call by dialing the IP addresses or an alias Server Registration is not required Endpoint Type H 323 Phone we Endpoint Mode Direct Mode e Direct Mode Gatekeeper Mode Release 1 0 13 Web Configuration HT 342 User Manual 2 Config Mode The device supports two modes Single Config Config by Line and Config by Group Single Config offers only one phone number and Gatekeeper configuration for both FXO lines Call Settings Endpoint Type H Advanced Settings hgt Endpoint Mode Gatekeeper Mode w Media Settings gt Config Mode Single Contig 4 Phone Number GateWay Prefix Display Name H 323 ID Gatekeeper Address ll _ Enable Authentication Config by Line allows each FXO line to have its own configuration of the following parameters phone number H 323 ID Gatekeeper Address Signaling Encryption method H 235 Auth Dial Plan and Fax Line Call Settings Endpoint Type H 323 Phone Advanced Settings Endpoint Mode Gatekeeper Mode Media Settings gt O Line1 OLine2 Phone Number H 323 1D batekeeper Address fs LJ Enable VOS AVS signaling Encryption O H 235 A
31. ut Network Tones INFO Server China Phone Code Preference Languages English Network Tones China Time Zone L PSTN Has Live Reversal Time Server pool ntp org DTMF Min Detect TimeGap Auto provision O Enable Disable provision interval J 1 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 kP IX Once the language change is saved it does not take effect until the device is rebooted Language ta a To change the display language immediately you can select the language icon as shown below However this does not change the default language EPH 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 date and time information It 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
32. uth Fax Line Config by Group allows up to 4 configuration groups for the two FXO lines Each configuration group includes the following parameters phone number H 323 ID Gatekeeper Address GateWay Prefix Signaling Encryption method H 235 Auth Dial Plan Activated Lines and Fax Line The Line Parameter specifies which FXO lines are Release 1 0 14 Web Configuration HT 342 User Manual included in the group Each FXO line can be assigned to any of the groups Call Settings Endpoint Type Advanced Settings Endpoint Moce Media Settings Config Mode T Group 1 Group OGroup3 Group 4 E Enable VOS AVS Signaling Encryption 0 H 235 Auth Activated Lines in Group 1 OL La Fax Line gt Config by 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 H 323 ID This parameter is specified in the H 323 protocol It is an identifier containing an alphanumeric string Some gatekeepers may use this ID for authentication 6 GateWay Prefix This assigns a prefix for routing PSTN calls via the HT 342 automatically Dial the prefix and then a PSTN number will result the call to be routed and dialed out via the corresponding FXO port 7 Gatekeeper Address This assigns the location of the Gatekeepe
33. xed 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 Release 1 0 22 Web Configuration HT 342 User Manual 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 supported Media QoS 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 Mone we Ri4 Encryption Media NAT Traversal ET263 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
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