Grandstream Networks, Inc.
Contents
1. 16 16 16 Transfer itte 16 Attended 16 3 WAY CONFERENCING eseeesesssesenn e ececeeecuseueeesessesueeaueeseseceeseuaeaaeesseseeeueeauausaeeeeeseeaneneeseees 17 FA Pe OR PO 17 CALL FEATURES in ee ean se pee ne mee 19 CONFIGURATION 21 CONFIGURING THE HT802 THROUGH VOICE 21 CONFIGURING THE HT802 VIA WEB BROWSER 21 Access the Web Configuration Menu esses sienne innen tnnt 21 IMPORTANT SETTINGS ic 22 INA TS CUO 22 DIM Methods ep pv te te dee bu ape Re 23 FIRMWARE VERSION 1 0 0 5 HT802 USER MANUAL Page 1 of 46 http ArtTel ru ustream Innovative IP Voice amp Video Preferred VOCODER 23 SAVING THE CONFIGURATION CHANGES2. enne entente nnne 38 REBOOTING THE HT802 FROM 38 SOFTWARE UPGRADE 39 FIRMWARE UPGRADE THROUGH 39 INSTRUCTIONS FOR UPLOAD FROM LOCAL 40 INSTRUCTIONS FOR LOCAL FIRMWARE UPGRADE USING TFTP SERVER 40 CONFIGURATION FILE DOWNLOAD essere nennen 40 FIRMWARE AND CONFIGURATION FILE PREFIX AND 0404400 0 41 MANAGING FIRMWARE AND CONFIGURATION FILE 41 RESTORE FACTORY DEFAULT SETTING 43 FACTORY RESE e 43 Reset 43 IVR Command Em 43 Reset from web interface Reset 2 44 FIRMWARE VERSION 1 0 0 5 HT802 USER MANUAL Page 2 of 46 http ArtTel ru stream Innovative IP Voice amp Video TABLE OF FIGURES HT802 USER MANUAL 2 aaa a a a a 8 TABLE OF TABLES HT802 USER MANUAL TABLE 1 DEFINISIONS OF THE HT802 2 9 TABLE 2 BASIC DEFINITIONS OF THE HT802 LEDS 9 TABLE 4 HT802 SOFTWARE FEATURES ccccccccccceeeeeecceeeeceaueuseseceseeeueaueessceeeceeeaueaseeeeeu
3. http www grandstream com general gs provisioning guide public pdf FIRMWARE AND CONFIGURATION FILE PREFIX AND POSTFIX Firmware Prefix and Postfix allows device to download the firmware name with the matching Prefix and Postfix This makes it the possible to store ALL of the firmware with different version in one single directory Similarly Config File Prefix and Postfix allows device to download the configuration file with the matching Prefix and Postfix Thus multiple configuration files for the same device can be stored in one directory In addition when the field Check New Firmware only when F W pre suffix changes is set to Yes the device will only issue firmware upgrade request if there are changes in the firmware Prefix or Postfix MANAGING FIRMWARE AND CONFIGURATION FILE DOWNLOAD When Automatic Upgrade is set Yes every the auto check will be done in the minute specified in this field If set to daily at hour 0 23 Service Provider can use P193 Auto Check Interval to have the devices do a daily check at the hour set in this field with either Firmware Server or Config Server If set to weekly on day 0 6 the auto check will be done in the day specified in this field This allows the device periodically check if there are any new changes need to be taken on a scheduled time By defining different intervals in P193 for different devices Server Provider can spread the Firmware or Configuration File download in min
4. conduct local firmware upgrade A free windows version TFTP server is available for download from http support solarwinds net updates New customerFree cfm Our latest official release can be downloaded from http www grandstream com support firmware INSTRUCTIONS FOR UPLOAD FROM LOCAL DIRECTORY 1 Download the firmware file from Grandstream web site 2 Unzip it and copy the file in to a folder in your PC 3 From the HT802 web interface Advanced Settings page you can browse your hard drive and select the folder you previously saved the file HT802fw bin 4 Click Upload Firmware and wait few minutes until the new program is loaded Note Always check the status page to see that the program version has changed INSTRUCTIONS FOR LOCAL FIRMWARE UPGRADE USING TFTP SERVER 1 Unzip the file and put all of them under the root directory of the TFTP server 2 Putthe PC running the TFTP server and the HT802 device in the same LAN segment 3 Please go to File gt Configure gt Security to change the TFTP server s default setting from Receive Only to Transmit Only for the firmware upgrade 4 Start the TFTP server in the phone s web configuration page 5 Configure the Firmware Server Path with the IP address of the PC 6 Update the change and reboot the unit End users can also choose to download the free HTTP server from http httpd apache org or use Microsoft IIS web server CONFIGURATION FILE DOWNLOAD Grandstre
5. FIRMWARE VERSION 1 0 0 5 http ArtTel ru HT802 USER MANUAL Page 31 of 46 Enable RTCP Refer to Use Target Contact Transfer on Conference Hang up Disable Bellcore Style 3 Way Conference Remove OBP from Route Header Support SIP Instance ID Validate incoming SIP message Check SIP User ID for incoming INVITE Allow Incoming SIP Messages from SIP Proxy Only SIP T1 Timeout SIP T2 Interval DTMF Payload Type Preferred DTMF method in listed order Disable DTMF Negotiation Send Hook Flash Event Enable Call Features andstream Innovative IP Voice amp Video NAT Default is Yes Enable Disable RTCP to provides out of band statistics and control information for RTP session Default is No If set to YES then for Attended Transfer the Refer To header uses the transferred target s Contact header information Default is No In which case if the conference originator hangs up the conference will be terminated When option YES is chosen originator will transfer other parties to each other so that B and C can choose to either continue the conversation or hang up Default is No you can make a Conference by pressing Flash key If set to Yes you need to dial 23 second callee number Default is No When option YES is chosen the Out Bound Proxy will be removed from Route header Default is Yes If set to Yes the contact header in REGISTER request will contain SIP Instance ID as defin
6. QoS Transport Protocol DTMF Method IP Signaling Provisioning Control Device Management Dial Plan Caller ID Call Handling Features ndstream Innovative IP Voice amp Video Yes DHCP Client only Yes T 38 compliant Group 3 Fax Relay up to 14 4kpbs and auto switch to G 711 for Fax Pass through Fax Datapump V 17 V 19 V 27ter V 29 for T 38 fax relay Diffserve TOS 802 1 P Q VLAN tagging RTP RTCP Flexible DTMF transmission method user interface of In audio RFC2833 and or SIP Info SIP RFC 3261 TFTP HTTP HTTPS TR 069 XML TLS SIPS SIP over TCP TLS Web interface or via secure encrypted AES or non encrypted central configuration file for mass deployment using Grandstream binary file or xml format Auto manual provisioning System or via built in IVR NAT friendly remote software upgrade via TFTP HTTP HTTPS for deployed devices including behind firewall NAT Syslog support Full support of TR 069 management protocol and Telnet access Yes Bellcore Type 1 amp 2 ETSI BT NTT and DTMF based CID Caller ID display or block Call waiting caller ID Call waiting flash Call transfer hold forward 3 way conferencing message waiting Do Not Disturb DND call return service HARDWARE SPECIFICATION The table below lists the Hardware and Technical specification of HT802 Telephone Interfaces Network Interface LED Indicators Factory Reset Button Universal Switching Power Adaptor Environmenta
7. 15 to choose firmware and configuration upgrade protocol User can choose between TFTP and HTTP FIRMWARE UPGRADE MODE Select voice menu option 17 to choose firmware upgrade mode among the following three options 1 Always check 2 check when pre suffix changes and 3 never upgrade CONFIGURING THE HT802 VIA WEB BROWSER HT802 has an embedded Web server that will respond to HTTP GET POST requests It also has embedded HTML pages that allow users to configure the HT802 through a web browser such as Microsoft s IE AOL s Netscape or Mozilla Firefox installed on Windows or Unix OS Macintosh OS is not included ACCESS THE WEB CONFIGURATION MENU 1 Find the IP address of the HT802 using voice prompt menu option 02 FIRMWARE VERSION 1 0 0 5 HT802 USER MANUAL Page 21 of 46 http ArtTel ru ustream Innovative IP Voice amp Video 2 Open a web browser type the IP address You will see the log in page of the device Note IVR announces 12 digits IP address you need to strip out the leading 0 in the IP address For ex IP address 192 168 001 014 you need to type in http 192 168 1 14 in the web browser Once the HTTP request is entered and sent from a web browser the user will see log in screen There are two default passwords for the login page User Level Password Web pages allowed End User Level 123 Only Status and Basic Settings Administrator Level Check with your service provid
8. 20ms or 30ms Defines payload type for iLBC Default value is 97 The valid range is between 96 and 127 Default is No VAD allows detecting the absence of audio and conserve bandwidth by preventing the transmission of silent packets over the network Default is No When set to Yes the device will change the destination to send RTP packets to the source IP address and port of the inbound RTP packet last received by the device T 38 Auto Detect FoIP by default or Pass Through must use codec PCMU PCMA Default is Enabled It makes the unit send out the re INVITE for T 38 or Fax Pass Through if a fax tone is detected FIRMWARE VERSION 1 0 0 5 http ArtTel ru HT802 USER MANUAL Page 36 of 46 Jitter Buffer Type Jitter Buffer Length SRTP Mode SLIC Setting Caller ID Scheme Polarity Reversal Loop Current Disconnect Loop Current Disconnect Duration Hook Flash Timing On Hook Timing Gain Disable Line Echo Canceller LEC Outgoing Call Duration Limit Andstream Innovative IP Voice amp Video Select either Fixed or Adaptive based on network conditions Default is Adaptive Select Low Medium or High based on network conditions Default is Medium e High initial 200ms min 40ms max 600ms Note not all vocoders can meet the high requirement e Medium initial 100ms 20ms max 200ms e Low initial 50 5 min 10ms 100ms This option defines different implementation of suppor
9. VERSION 1 0 0 5 HT802 USER MANUAL Page 4 of 46 http ArtTel ru dstream Innovative IP Voice amp Video WELCOME Thank you for purchasing Grandstream s HT802 the affordable feature rich Analog Telephone Adaptor Grandstream HandyTone802 is a new addition to the popular HandyTone ATA product family It features the rich audio quality a broad range of voice codecs and functionality including one independent SIP account per FXS port This manual will help you learn how to operate and manage your HandyTone70X Analog Telephone Adaptor and make the best use of its many upgraded features including simple and quick installation 3 way conferencing direct IP IP Calling and new provisioning support among other features This HT802 is very easy to manage and configure and is specifically designed to be an easy to use and affordable VoIP solution for both the residential user and the teleworker SAFETY COMPLIANCES The HT802 phone complies with FCC CE and various safety standards The HT802 power adaptor is compliant with UL standard Only use the universal power adapter provided with the HT802 package The manufacturers warranty does not cover damages to the phone caused by unsupported power adaptors WARRANTY If you purchased your HT802 from a reseller please contact the company where you purchased your device for replacement repair or refund If you purchased the product directly from Grandstream contact your Grandstream Sales and
10. administrator can configure the Advanced Settings page Password field is purposely left blank for security reasons after clicking update and saved The maximum password length is 25 characters Set values for Diff Serv or Precedence value for both SIP and RTP Set values for 802 1Q VLAN Tag 0 4094 802 1p priority value 0 7 802 1p priority value NATed traffic 0 7 IP address or Domain name of the STUN server This parameter specifies how often the HT802 sends a blank UDP packet to the SIP server in order to keep the hole on the NAT open Default is 20 seconds Minimum value is 20 seconds Use STUN keep alive to detect WAN side network problems If keep alive request does not yield any response for configured number of times the device will restart the TCP IP stack If the STUN server does not respond when the device boots up the feature is disabled Default setting is No Use DNS to detect WAN side network problems Default setting is Yes Enable Disable telnet server on device Enables HT802 to download firmware or configuration file through either the TFTP HTTP or HTTPS server This is the IP address of the configured TFTP server If selected and it is non zero or not blank the HT802 retrieves the new configuration file or new code image from the specified TFTP server at boot time After 5 attempts the system will timeout and will start the boot process using the existing code image in the Flash memory I
11. amp Video Set this to Yes when gateway is behind firewall on a private network DTMF METHODS DTMF Settings are in FXS portX Profilex page DTMF in audio DTMF via RTP RFC2833 DTMF via SIP INFO Set priority of DTMF methods according to your preference This setting should be based on your server DTMF setting PREFERRED VOCODER CODEC The HT802 supports a broad range of voice codecs Under Profile web pages choose your preferred order of different codecs or G711y a e G729 A B G723 1 e 0726 16 24 32 40 e iLBC End User Password Web Port IP Address DHCP hostname TABLE 7 BASIC SETTINGS Password to access the Web Configuration Menu This field is case sensitive with a maximum length of 25 characters By default HTTP uses port 80 This field is for customizable web port There are two modes to operate the HT802 dynamically assigned via DHCP all the field values for the Static IP mode are not used even though they are still saved in the Flash memory The HT802 acquires its IP address from the first DHCP server it discovers from the LAN it is connected Statically configured as configure the IP address Subnet Mask Default Router IP address Default is blank This option specifies the name of the client This field is optional but may be required by some Internet Service Providers FIRMWARE VERSION 1 0 0 5 HT802 USER MANUAL Page 23 of 46 http A
12. with a single firmware file by browsing and loading the file from your computer local directory TABLE 10 ACCOUNT SETTINGS When set to Yes the FXS port or Profile for HT704 is activated SIP Server s IP address or Domain name provided by VoIP service provider Failover SIP Server s IP address or Domain name in case primary server does not respond Default is No If set to yes it will register to Primary Server if registration with Failover server expires IP address or Domain name of Outbound Proxy or Media Gateway or Session Border Controller Used by HT802 for firewall or NAT penetration in different network environments If symmetric NAT is detected STUN will not work and ONLY outbound proxy can correct the problem User can select UDP or TCP or TLS Default is UDP This parameter defines whether or not the HT802 NAT traversal mechanism is activated If activated by choosing Yes and a STUN server is also specified then the HT802 performs according to the STUN client specification Using this mode the embedded STUN client will detect if and what type of firewall NAT If the detected NAT is a Full Cone Restricted Cone or a Port Restricted Cone the HT802 will use its mapped public IP address and port in all of its SIP and SDP messages If the NAT Traversal field is set to Yes with no specified STUN server the HT802 will periodically every 20 seconds or so send a blank UDP packet with no payload data to the S
13. IP server to keep the hole on the NAT open User account information provided by VoIP service provider ITSP Usually in the form of digit similar to phone number or actually a phone number SIP service subscribers Authenticate ID used for authentication Can be identical to or different from SIP User ID FIRMWARE VERSION 1 0 0 5 HT802 USER MANUAL Page 30 of 46 http ArtTel ru Authenticate Password Name DNS Mode Tel URI SIP Registration Unregister on Reboot Outgoing Call without Registration Register Expiration SIP Registration Failure Retry Wait Time Local SIP port Local RTP port Use Random SIP Port Use Random RTP Port andstream Innovative IP Voice amp Video SIP service subscriber s account password SIP service subscriber s name for Caller ID display One from the 3 modes are available for DNS Mode configuration A Record for resolving IP Address of target according to domain name SRV DNS SRV resource records indicates how to find services for various protocols NAPTR SRV Naming Authority Pointer according to RFC 2915 One mode can be chosen for the client to look up server The default value is A Record The default setting is Disabled If the phone has an assigned PSTN Number this field should be set to User Phone then a User Phone parameter will be attached to the From header in the SIP request to indicate the E 164 number If s
14. Service Representative for a RMA Return Materials Authorization number before you return the product Grandstream reserves the right to remedy warranty policy without prior notification Caution Changes or modifications to this product not expressly approved by Grandstream or operation of this product in any way other than as detailed by this User Manual could void your manufacturer warranty Please do not use a different power adaptor with the HT802 as it may cause damage to the products and void the manufacturer warranty Reproduction or transmittal of the entire or any part in any form or by any means electronic or print for any purpose is not permitted without the express written permission of Grandstream Networks Inc FIRMWARE VERSION 1 0 0 5 HT802 USER MANUAL Page 5 of 46 http ArtTel ru ustream Innovative IP Voice amp Video This page is intentionally left bank FIRMWARE VERSION 1 0 0 5 HT802 USER MANUAL Page 6 of 46 http ArtTel ru ustream Innovative IP Voice amp Video CONNECT YOUR HT802 Connecting the HT802 is easy Before you begin please verify the contents of the HT802 package EQUIPMENT PACKAGING The HT802 ATA package contains e One HT802 Main Case e One Universal Power Adaptor e One Ethernet Cable e One Vertical Stand Only on HT702 and HT704 Packages CONNECTING THE HT802 The HT802 is designed for easy configuration and easy installation Configure the HT802 following t
15. USER MANUAL Page 15 of 46 http ArtTel ru dstream Innovative IP Voice amp Video b If the target IP address port is 192 168 1 20 5062 then the dialing convention would be 47 or Voice Prompt with option 47 then 192 168 0 160 5062 followed by pressing the key if it is configured as a send key or wait for 4 seconds NOTE When completing direct IP call the Use Random SIP RTP Port should set to NO You cannot make direct IP calls between FXS1 to FXS2 since they are using same IP CALL HOLD Place a call on hold by pressing the flash button on the analog phone if the phone has that button Press the flash button again to release the previously held Caller and resume conversation If no flash button is available use hook flash toggle on off hook quickly You may drop a call using hook flash CALL WAITING Call waiting tone 3 short beeps indicates an incoming call if the call waiting feature is enabled Toggle between incoming call and current call by pressing the flash button First call is placed on hold Press the flash button to toggle between two active calls CALL TRANSFER BLIND TRANSFER Assume that call Caller A and B are in conversation A wants to Blind Transfer B to C 3 Caller A presses FLASH on the analog phone to hear the dial tone 4 Caller A dials 87 then dials caller C s number and then or wait for 4 seconds 5 Caller A will hear the dial
16. USER MANUAL Page 38 of 46 http ArtTel ru dstream Innovative IP Voice amp Video SOFTWARE UPGRADE Software upgrade can be done via either TFTP or HTTP The corresponding configuration settings are in the ADVANCED SETTINGS configuration page FIRMWARE UPGRADE THROUGH TFTP HTTP HTTPS To upgrade via TFTP or HTTP HTTPS the Firmware Upgrade and Provisioning upgrade via field needs to be set to TFTP HTTP or HTTPS respectively Firmware Server Path needs to be set to a valid URL of a TFTP or HTTP Server server name can be in either FQDN or IP address format Here are examples of some valid URL e g firmware mycompany com 6688 Grandstream 1 0 5 2 e g firmware grandstream com NOTES e Firmware upgrade server IP address format can be configured via IVR Please refer to the CONFIGURATION GUIDE section for instructions If the server is in FQDN format it must be set via the web configuration interface e Grandstream recommends end user use the Grandstream HTTP server Its address can be found at http Awww grandstream com support firmware Currently the HTTP firmware server IP address is firmware grandstream com For large companies we recommend to maintain their own TFTP HTTP HTTPS server for upgrade and provisioning procedures e Once a Firmware Server Path is set user needs to update the settings and reboot the device If the configured firmware server is found and a new code image is available the HT802 w
17. address 4 Wait 15 seconds and device will automatically reboot and restore factory settings Encode the MAC Address 1 Locate the MAC address of the device It is the 12 digit HEX number on the bottom of the unit 2 Key in the MAC address Use the following mapping 0 9 0 9 A 22 press the 2 key twice A will show on the LCD B 222 FIRMWARE VERSION 1 0 0 5 HT802 USER MANUAL Page 43 of 46 http ArtTel ru stream Innovative IP Voice amp Video C 2222 D 33 press the 3 key twice D will show on the LCD E 333 F 3333 For example if the MAC address is 000b8200e395 it should be keyed in as 0002228200333395 RESET FROM WEB INTERFACE RESET TYPE 1 From the Advanced Settings Page user can select three types e Full Reset This will make a full reset e ISP Data This will reset only the basic settings like IP mode PPPoE and Web port e VOIP Data This will reset only the data related with a service provider like SIP server sip user ID provisioning and others NOTE 1 Factory Reset will be disabled if the Lock keypad update is set to Yes 2 f the HT802 was previously locked by your local service provider pressing the RESET button will only restart the unit The device will not return to factory default settings FIRMWARE VERSION 1 0 0 5 HT802 USER MANUAL Page 44 of 46 http ArtTel ru dstream Innovative IP Voice amp Video FCC Compliance Statement This dev
18. ading from left to right 1900x prevents dialing any number started with 1900 lt 1617 gt 2 9 xxxxxx allows dialing to local area code 617 numbers by dialing 7 numbers and 1617 area code will be added automatically 1 2 9 xx 2 9 xxxxxx allows dialing to any US Canada Number with 11 digits length 011 2 9 x allows international calls starting with 01 1 3469 11 allow dialing special and emergency numbers 311 411 611 and 911 Note In some cases user wishes to dial strings such as 123 to activate voice mail or other application provided by service provider In this case should be predefined inside dial plan feature and the Dial Plan should be x Default is No When set to Yes a SUBSCRIBE for Message Waiting Indication will be sent periodically Default is No If this parameter is set to Yes the From header along with Privacy and P Asserted Identity headers in outgoing INVITE message will be set to anonymous blocking Caller ID Default is No If set to Yes incoming calls with anonymous Caller ID will be rejected with 486 Busy message Default is Standard Choose the selection to meet some special requirements from Softswitch vendors Grandstream implemented SIP Session Timer The session timer extension enables SIP sessions to be periodically refreshed via a SIP request UPDATE or re INVITE Once the session interval expires if there is no refresh via a UPDATE or re INVITE message the ses
19. am SIP Device can be configured via Web Interface as well as via Configuration File through TFTP or HTTP HTTPS Config Server Path is the TFTP or HTTP HTTPS server path for configuration FIRMWARE VERSION 1 0 0 5 HT802 USER MANUAL Page 40 of 46 http ArtTel ru ustream Innovative IP Voice amp Video file It needs to be set to a valid URL either in FQDN or IP address format The Config Server Path can be same or different from the Firmware Server Path A configuration parameter is associated with each particular field in the web configuration page A parameter consists of a Capital letter P and 1 to 3 Could be extended to 4 in the future digit numeric numbers i e P2 is associated with Admin Password in the ADVANCED SETTINGS For a detailed parameter list please refer to the corresponding firmware release configuration template When a Grandstream device boots up or reboots it will issue a request for a configuration file where is the MAC address of the device i e cfg000b820102ab In addition device will also requests a XML configuration file cfgxxxxxxxxxxxx xml If the download of cfgxxxxxxxxxxxx xml is not successful the provision program will issue a request for a generic configuration file cfg xml Configuration file name should be in lower case letters For more details on XML provisioning please refer to
20. calls 31 Send Caller ID for all subsequent calls 47 Direct IP Calling Dial 47 IP address dial tone is played in the middle Detail see Direct IP Calling section on page 12 50 Disable Call Waiting for all subsequent calls 51 Enable Call Waiting for all subsequent calls 67 Block Caller ID per call Dial 67 number No dial tone is played in the middle 82 Send Caller ID per call Dial 82 number No dial tone is played in the middle 69 Call Return Service Dial 69 and the phone will dial the last incoming phone number received 70 Disable Call Waiting per call Dial 70 number No dial tone is played in the middle 71 Enable Call Waiting per call Dial 71 number No dial tone is played in the middle 72 Unconditional Call Forward Dial 72 and then the forwarding number followed by Wait for dial tone and hang up dial tone indicates successful forward 73 Cancel Unconditional Call Forward To cancel Unconditional Call Forward dial 73 wait for dial tone then hang up 74 Enable Paging Call Dial 74 and then the destination phone number you want to page 78 Enable Do Not Disturb DND When enabled all incoming calls are rejected 79 Disable Do Not Disturb DND When disabled incoming calls are accepted 87 Blind Transfer 90 Busy Call Forward Dial 90 and then the forwarding number follo
21. ded as part of number Dial Plan Dial Plan Rules 1 Accept Digits 1 2 3 4 5 6 7 8 9 0 A a B b C c D d 2 Grammar x any digit from 0 9 e xx atleast 2 digits number e xx at least 2 digit number exclude e 3 5 any digit of 3 4 or 5 e 147 any digit 1 4 or 7 e 2 011 replace digit 2 with 011 when dialing e lt 1 gt add a leading 1 to all numbers dialed vice versa will remove a 1 from the number dialed e or Example 1 369 11 1617xxxxxxx Allow 311 611 911 and any 11 digit numbers with leading digits 1617 Example 2 71900 lt 1617 gt xxxxxxx Block any number of leading digits FIRMWARE VERSION 1 0 0 5 HT802 USER MANUAL Page 34 of 46 http ArtTel ru Subscribe for MWI Send Anonymous Anonymous Call Rejection Special Feature Session Expiration Min SE Caller Request Timer andstream Innovative IP Voice amp Video 1900 and add prefix 1617 for any dialed 7 digit numbers Example 3 1xxx 2 9 xxxxxx lt 2 011 gt x Allow any combinations of numbers with 11 digits which has a leading digit 1 but 5th digit cannot be 0 or 1 Or any length of numbers with a minimum of 2 digits beginning with 2 with the leading digit replaced with 011 3 Default Outgoing x Example of a simple dial plan used in a Home Office in the US 1900x lt 1617 gt 2 9 xxxxxx 1 2 9 xx 2 9 xxxxxx 011 2 9 x 3469 11 Explanation of example rule re
22. e dropped out 6 If A hangs up the conference will be terminated for all three parties when configuration Transfer on Conference Hang up is set to No If the configuration is set to Yes A will transfer B to C so that B and C can continue the conversation FAX SUPPORT HT802 supports FAX in two modes 1 T 38 Fax over IP and 2 fax pass through T 38 is the preferred method because it is more reliable and works well in most network conditions If the service provider supports T 38 please use this method by selecting Fax mode to be T 38 default If the service provider does not support T 38 pass through mode can be used FIRMWARE VERSION 1 0 0 5 HT802 USER MANUAL Page 17 of 46 http ArtTel ru ustream Innovative IP Voice amp Video This page is intentionally left bank FIRMWARE VERSION 1 0 0 5 HT802 USER MANUAL Page 18 of 46 http ArtTel ru andstream Innovative IP Voice amp Video CALL FEATURES The HT802 supports all the traditional and advanced telephony features TABLE 6 HT802 CALL FEATURES CALL FEATURES 02 Forcing a Codec per call 027110 027111 02723 G723 02729 G729 0272616 G726 r16 0272624 G724 r24 0272632 G726 r32 0272640 G726 r40 027201 iLBC 03 Disable LEC per call Dial 03 number No dial tone is played in the middle 16 Enable SRTP 17 Disable SRTP 30 Block Caller ID for all subsequent
23. e the reset type you will have to click the reset button for it to take effect In addition to the Basic Settings configuration page end users also have access to the Device Status page TABLE 8 STATUS PAGE MAC Address The device ID in HEX format This is very important ID for ISP troubleshooting The Mac address will appear in this place The MAC address will be used for provisioning and can be found on the label coming with original box and on the label located on the bottom panel of the device FIRMWARE VERSION 1 0 0 5 HT802 USER MANUAL Page 24 of 46 http ArtTel ru IP Address Product Model Hardware Version Software Version Software Status System Up Time PPPoE Link Up NAT Port Status Core Dump Provision Andstream Innovative IP Voice amp Video This field shows the IP address of the HT802 This field contains the product model info This field shows the hardware revision of the unit and the part number Program This is the main software release This number is always used for firmware upgrade Current release is 1 0 3 1 Boot and Loader are seldom changed Bootloader current version is 1 0 0 7 Core current version 1 0 3 1 Base current version is 1 0 3 1 CPE gs_cpe version number information Shows software running status and memory status Shows system up time since the last reboot Indicates whether the PPPoE connection is up if the HT802 is connected to DSL modem This filed
24. ed in IETF SIP Outbound draft Default is No If set to yes all incoming SIP messages will be strictly validated according to RFC rules If message will not pass validation process call will be rejected Default is No Check the incoming SIP User ID in Request URI If they don t match the call will be rejected If this option is enabled the device will not be able to make direct IP calls Default is No Check the incoming SIP messages If they don t come from the SIP proxy they will be rejected If this option is enabled the device will not be able to make direct IP calls T1 is an estimate of the round trip time between the client and server transactions If the network latency is high select larger value for more reliable usage Default is 0 5 Sec Maximum retransmission interval for non INVITE requests and INVITE responses Default is 4 Sec Sets the payload type for using RFC2833 Default is 101 The HT802 supports up to 3 different DTMF methods including in audio via RTP RFC2833 and via Sip Info using SIP INFO messages The user can configure DTMF method in a priority list Default is No If set to yes use above DTMF order without negotiation Default is No If set to yes flash will be sent as DTMF event Default is Yes If Yes call features using star codes will be supported locally FIRMWARE VERSION 1 0 0 5 http ArtTel ru HT802 USER MANUAL Page 32 of 46 Offhook Auto Dial Offhook Auto Dial De
25. efix numbers DIRECT IP CALLS Direct IP calling allows two parties that is a FXS Port with an analog phone and another VoIP Device to talk to each other in an ad hoc fashion without a SIP proxy Elements necessary to completing a Direct IP Call 1 Both HT802 and other VoIP Device have public IP addresses or 2 Both HT802 and other VoIP Device are on the same LAN using private IP addresses or 3 Both HT802 and other VoIP Device can be connected through a router using public or private IP addresses with necessary port forwarding or DMZ HT802 supports two ways to make Direct IP Calling Using IVR 1 Pick up the analog phone then access the voice menu prompt by dial 2 Dial 47 to access the direct IP call menu 3 Enter the IP address after the dial tone and voice prompt Direct IP Calling Using Star Code 1 Pick up the analog phone then dial 47 2 Enter the target IP address Note NO dial tone will be played between step 1 and 2 Destination ports can be specified using encoding for followed by the port number Examples of Direct IP Calls a Ifthe target IP address is 192 168 0 160 the dialing convention is 47 or Voice Prompt with option 47 then 192 168 0 160 followed by pressing the 4 key if it is configured as a send key or wait 4 seconds In this case the default destination port 5060 is used if no port is specified FIRMWARE VERSION 1 0 0 5 HT802
26. er Browse all pages The password is case sensitive with maximum length of 25 characters The factory default password for End User and administrator is 123 and admin respectively Only an administrator can access the ADVANCED SETTING FXS PORTs configuration pages NOTE HT802 has factory configuration with random administrator password Check with your service provider for more information IMPORTANT SETTINGS The end user must configure the following settings according to the local environment NOTE Most settings on the web configuration pages are set to the default values NAT SETTINGS If you plan to keep the Handy Tone within a private network behind a firewall we recommend using STUN Server The following three 3 settings are useful in the STUN Server scenario 1 STUN Server under Advanced Settings webpage Enter a STUN Server IP or FQDN that you may have or look up a free public STUN Server on the internet and enter it on this field If using Public IP keep this field blank 2 Use Random SIP RTP Ports under Advanced Settings webpage This setting depends on your network settings Generally if you have multiple IP devices under the same network it should be set to Yes If using a Public IP address set this parameter to No 3 NAT Traversal under the FXS Profile web page FIRMWARE VERSION 1 0 0 5 HT802 USER MANUAL Page 22 of 46 http ArtTel ru dstream Innovative IP Voice
27. erver supports TEL URI format then this option needs to be selected Controls whether the HT802 needs to send REGISTER messages to the proxy server The default setting is Yes Default is No If set to Yes the SIP user s registration information will be cleared on reboot Default is No If set to Yes user can place outgoing calls even when not registered if allowed by Internet Telephone Service Provider but is unable to receive incoming calls This parameter allows the user to specify the time frequency in minutes the HT802 refreshes its registration with the specified registrar The default interval is 60 minutes or 1 hour The maximum interval is 65535 minutes about 45 days Retry registration if the process failed Default is 20 seconds Defines the local SIP port the HT802 will listen and transmit The default value for FXS port is 5060 Defines the local RTP RTCP port pair the HT802 will listen and transmit It is the base RTP port for channel 0 When configured channel 0 uses this value for RTP and the port_value 1 for its RTCP The default value for FXS port is 5004 Default is No This parameter forces the random generation of The local SIP ports when set to Yes This is usually necessary when multiple HT802 are behind the same NAT Default is No This parameter forces the random generation of the local RTP ports when set to Yes This is usually necessary when multiple HT802 are behind the same
28. eseeaueaseeeeseseeananeaeess 11 TABLE 5 HT802 HARDWARE AND TECHNICAL 12 TABLE 6 HT802 IVR MENU DEFINITIONS 13 TABLE 7 HT802 CALL FEATURE O 19 TABLE 8 BASIC 5 5 23 TABLE 9 STATUS rere eere recorded eve ee pee ever dc de dated 24 TABLE 10 ADVANCED 5 5 26 TABLE T1 ACCOUNT SETTINGS eet rte epe e 30 TABLE 12 HT704 FXS PORTS SETTINGS 0200 BRIE LBS CONFIGURATION GUI INTERFACE EXAMPLES HT802 USER MANUAL 1 SCREENSHOT OF ADVANCED USER CONFIGURATION PAGE 2 SCREENSHOT OF BASIC SETTINGS CONFIGURATION PAGE 3 SCREENSHOT OF FXS PORT CONFIGURATION 4 SCREENSHOT OF STATUS PAGE 5 SCREENSHOT OF LOGIN PAGE 6 SCREENSHOT OF REBOOT PAGE 7 SCREENSHOT OF REBOOTING PAGE FIRMWARE VERSION 1 0 0 5 HT802 USER MANUAL Page 3 of 46 http ArtTel ru stream Innovative IP Voice amp Video GNU GPL INFORMATION HT802 firmware contains third party software licensed under the GNU General Public License GPL Grandstream uses software under the specific terms of the GPL Please see the GNU General Public License GPL for the exact terms and conditions of the license Grandstream GNU GPL related source code can be downloaded from Grandstream web site from http www grandstream com support fag gnu FIRMWARE
29. f a TFTP server is configured and a new code image is retrieved the new downloaded image is saved into the Flash memory Note Please do NOT interrupt the TFTP upgrade process especially the power supply as this will damage the device Depending on the local network this process can take up FIRMWARE VERSION 1 0 0 5 http ArtTel ru HT802 USER MANUAL Page 26 of 46 Andstream Innovative IP Voice amp Video to 15 or 20 minutes Via HTTP HTTPS The URL for the HTTP HTTPS server used for firmware upgrade and configuration via Server HTTP For example http provisioning mycompany com 6688 Grandstream 1 0 0 67 6688 is the specific TCP port where the HTTP or HTTPS server is listening it can be omitted if using default port 80 Note If Auto Upgrade is set to No HT802 will only do HTTP HTTPS download once at boot up Firmware Server Path IP address or domain name of firmware server Config Server Path IP address or domain name of configuration server XML Config File The password used for encrypting the XML configuration file using OpenSSL Password This is required for the phone to decrypt the encrypted XML configuration file HTTP HTTPS User The user name needed to authenticate with the HTTP HTTPS server Name HTTP HTTPS The password needed to authenticate with the HTTP HTTPS server Password Firmware File Prefix Default is blank If configured HT802 will request firmware file with the prefix This setting is
30. he directions in the Configuration section of this manual 1 Insert a standard RJ11 telephone cable into the Phone port and connect the other end of the telephone cable to a standard touch tone analog telephone 2 Insert the Ethernet cable into the Internet or LAN port of the HT802 and connect the other end of the Ethernet cable to an uplink port a router or a modem etc 3 Insert the power adapter into the HT802 and connect it to a wall outlet The HT802 Analog Telephone Adaptor is an all in one VoIP integrated device designed to be a total solution for networks providing VoIP services The HT802 VoIP features and functions are available using a regular analog telephone FIRMWARE VERSION 1 0 0 5 HT802 USER MANUAL Page 7 of 46 http ArtTel ru Innovative IP Voice amp Video Internet Port Mircro USB Power Telephone Jack 1 amp 2 RJ45 connector DC 5V 1A RJ11 connecter Power LED Indicator Telephone LED Telephone LED Indicator 1 Indicator 2 Internet LED Indicator FIGURE 1 DIAGRAM OF HT802 FIRMWARE VERSION 1 0 0 5 HT802 USER MANUAL Page 8 of 46 http ArtTel ru stream Innovative IP Voice amp Video TABLE 1 DEFINISIONS OF THE HT802 CONNECTORS Power Micro USB 5V 1A Internet Port RJ 45 Connect to network RESET Factory Reset button Press for 7 seconds to reset factory default settings Telephone Ports RJ 11 Used to connect to analog phones fax machines There a
31. ice complies with part 15 of the FCC Rules Operation is subject to the following two conditions 1 This device may not cause harmful interference and 2 this device must accept any interference received including interference that may cause undesired operation Important Any Changes or modifications not expressly approved by the party responsible for compliance could void the user s authority to operate the equipment Note This equipment has been tested and found to comply with the limits for a Class B digital device pursuant to part 15 of the FCC Rules These limits are designed to provide reasonable protection against harmful interference in a residential installation This equipment generates uses and can radiate radio frequency energy and if not installed and used in accordance with the instructions may cause harmful interference to radio communications However there is no guarantee that interference will not occur in a particular installation If this equipment does cause harmful interference to radio or television reception which can be determined by turning the equipment off and on the user is encouraged to try to correct the interference by one or more of the following measures Reorient or relocate the receiving antenna Increase the separation between the equipment and receiver Connect the equipment into an outlet on a circuit different from that to which the receiver is connected Consult the dealer or an experie
32. ill attempt to retrieve the new image files by downloading them into the HT802 s SRAM During this stage the HT802 s LEDs will blink until the checking downloading process is completed Upon verification of checksum the new code image will then be saved into the Flash If TFTP HTTP HTTPS fails for any reason e g TFTP HTTP HTTPS server is not responding there are no code image files available for upgrade or checksum test fails etc the HT802 will stop the TFTP HTTP HTTPS process and simply boot using the existing code image in the flash e Firmware upgrade may take as long as 15 to 30 minutes over Internet or just 5 minutes if it is performed on a LAN t is recommended to conduct firmware upgrade in a controlled LAN environment if possible For users who do not have a local firmware upgrade server Grandstream provides a NAT friendly TFTP server on the public Internet for firmware upgrade FIRMWARE VERSION 1 0 0 5 HT802 USER MANUAL Page 39 of 46 http ArtTel ru ustream Innovative IP Voice amp Video e Grandstream s latest firmware is available at http www grandstream com support firmware Oversea users are strongly recommended to download the binary files and upgrade firmware locally in a controlled LAN environment Note Alternatively the user can upload the firmware single file from his her computer using a local directory For more instructions see below Also the user can download a free TFTP or HTTP server and
33. indicates the type of NAT connection used by the HT802 Displays relevant information regarding the FXS port Port Hook Registration DND Forward Busy Delayed Forward Forward FXS On Hook Registered Yes 613 FXS port is registered with SIP Server FXS Port user has set Do Not Disturb FXS Port user has set his calls to be unconditionally forwarded to ext 613 FXS Port user has not set Busy or Delay call Forward Shows core dump if generated Keeps updating provisioning status Log in to the advanced user configuration page the same way as for the basic configuration page The password is case sensitive and the factory default password for Advanced User is admin FIRMWARE VERSION 1 0 0 5 http ArtTel ru HT802 USER MANUAL Page 25 of 46 Andstream Innovative IP Voice amp Video Advanced User configuration includes the end user configuration and the advanced configurations including a SIP configuration b Codec selection c NAT Traversal Setting and d other miscellaneous configuration Admin Password Layer 3 QoS Layer 2 QoS STUN Server is Keep alive interval Use STUN to detect network connectivity Use DNS to detect network connectivity Telnet Server Firmware Upgrade and Provisioning Via TFTP Server TABLE 9 ADVANCED SETTINGS This contains the password to access the Advanced Web Configuration page This field is case sensitive Only the
34. ing Dynamic IP Mode all IP address information comes from the DHCP server automatically after reboot The current WAN IP address is announced If using Static IP Mode enter 12 digit new IP address You need to reset the HT for the new IP address to take Effect Same as menu 02 Same as menu 02 Same as menu 02 Press 9 to move to the next selection in the list PCMU PCMA e G 726 e G 723 e G 729 Announces the address of the unit Announces current Firmware Server IP address Enter 12 digit new IP address Announces current Config Server Path IP address Enter 12 digit new IP address Upgrade protocol for firmware and configuration update Press 9 to toggle between TFTP HTTP HTTPS FIRMWARE VERSION 1 0 0 5 http ArtTel ru HT802 USER MANUAL Page 13 of 46 ustream Innovative IP Voice amp Video 16 Firmware Version Firmware version information 17 Firmware Upgrade Firmware upgrade mode Press 9 to toggle among the following three options always check check when pre suffix changes never upgrade 47 Direct IP Calling Enter the target IP address to make a direct IP call after dial tone See Make a Direct IP Call 86 Voice Mail Number of Voice Mails 99 RESET Press 9 to reboot the device Enter MAC address to restore factory default setting See Restore Factory Default Setting section Invalid Entry Automatically returns to main men
35. l TABLE 4 HT802 HARDWARE AND TECHNICAL SPECIFICATIONS 2 RJ11 FXS port 1 RJ45 10 100 Mb Base TX Full Duplex port with connectivity LEDs POWER INTERNET PHONE1 PHONE2 Yes Input AC 100 240 V 50 60 Hz 0 2A max Output DC 5V 1A UL FCC certified Operational 32 104 F or 0 40 C Storage 14 140 or 10 60 Humidity 10 9096 Non condensing FIRMWARE VERSION 1 0 0 5 http ArtTel ru HT802 USER MANUAL Page 12 of 46 ndstream Innovative IP Voice amp Video BASIC OPERATIONS UNDERSTANDING HT802 VOICE PROMPT HT802 has a built in voice prompt menu for simple device configuration The IVR menu and the LED button work with any of the FXS port Pick up the handset and dial to use the IVR menu Main Menu 01 02 03 04 05 07 10 13 14 15 TABLE 5 HT802 IVR MENU DEFINITIONS Enter a Menu Option DHCP Mode Static IP Mode IP Address IP address Subnet IP address Gateway IP address DNS Server IP address Preferred Vocoder MAC Address Firmware Server IP Address Configuration Server IP Address Upgrade Protocol Press for the next menu option Press to return to the main menu Enter 01 05 07 10 13 17 47 or 99 menu options Press 9 to toggle the selection If using Static IP Mode configure the IP address information using menus 02 to 05 If us
36. l parameter located on the FXS Port Configuration page If call volume is too low when using the FXS port ie the ATA is at user site adjust volume using the Rx Gain Level parameter under the FXS Port Configuration page If voice volume is too low at the other end user may increase the far end volume using the Tx Gain Level parameter under the FXS Port Configuration page Default is No If set to Yes LEC will be disabled per call base Recommended for FAX Data calls Default is 0 When set to 0 There will be no limit Time range 0 180 minutes FIRMWARE VERSION 1 0 0 5 http ArtTel ru HT802 USER MANUAL Page 37 of 46 ustream Innovative IP Voice amp Video Ring Tones This function lets you configure ring tone cadence preferences User has 10 choices The configuration completed in Distinctive Ring Tones block in the same page applies to ring tones cadences configured here SAVING THE CONFIGURATION CHANGES Click the Update button in the Configuration page to save the changes to the HT802 configuration The following screen confirms that the changes are saved Reboot or power cycle the HT802 to make the changes take effect REBOOTING THE HT802 FROM REMOTE Press the Reboot button at the bottom of the configuration menu to reboot the ATA remotely The web browser will then display a message window to confirm that reboot is underway Wait 30 seconds to log in again FIRMWARE VERSION 1 0 0 5 HT802
37. lay Proxy Require Use NAT IP Use SIP User Agent Header Distinctive Ring Tone Disable Call Waiting Disable Call Waiting Caller ID Disable Call Waiting Tone Disable Receiver Offhook Tone Disable Reminder Ring for On Hold Call Disable Visual MWI Andstream Innovative IP Voice amp Video This parameter allows users to configure a User ID or extension number that is automatically dialed when off hook Only the user part of a SIP address needs to be entered here The HT802 will automatically append the and the host portion of the corresponding SIP address The auto dial delay after off hook SIP Extension to notify SIP server that the unit is behind the NAT Firewall NAT IP address used in SIP SDP message Default is blank Configurable SIP User Agent Header Custom Ring Tone 1 to 3 with associate Caller ID when selected if Caller ID is configured then the device will ONLY uses this ring tone when the incoming call is from the Caller ID System Ring Tone is used for all other calls When selected but no Caller ID is configured the selected ring tone will be used for all incoming calls using the FXS port or Profile Distinctive ring tones can be configured not only for matching a whole number but also for matching prefixes In this case symbol star will be used For example if configured as 617 Ring Tone 1 will be used in case of call arrived from the area code 617 Any other incoming call will ri
38. llowing events 1 product model version on boot up INFO level NAT related info INFO level sent or received SIP message DEBUG level SIP message summary INFO level inbound and outbound calls INFO level registration status change INFO level negotiated codec INFO level Ethernet link up INFO level SLIC chip exception WARNING and ERROR levels Oo Noa 9 _ 2 memory exception ERROR level Vonage requested syslog style EXTRA DEBUG level The Syslog uses USER facility In addition to standard Syslog payload it contains the following components GS LOG device MAC address error code error message Example May 19 02 40 38 192 168 1 14 GS LOG 00 0b 82 00 a1 be 000 Ethernet link FIRMWARE VERSION 1 0 0 5 HT802 USER MANUAL Page 29 of 46 http ArtTel ru Send SIP Log Download Device Configuration Upload Firmware Account Active Primary SIP Server Failover SIP Server Prefer Primary SIP Server Outbound Proxy SIP transport NAT Traversal STUN SIP User ID Authenticate ID ndstream Innovative IP Voice amp Video is up Default is No If Enabled the HT802 will send a replicate of the SIP packets on the syslog Allows user to download and save a text file containing all the P values of each setting as configured at that point on the unit Note For Security Reasons all Passwords won t be Downloaded Allows the user to upgrade the firmware
39. nced radio TV technician for help FIRMWARE VERSION 1 0 0 5 HT802 USER MANUAL Page 45 of 46 http ArtTel ru
40. ndstream Innovative IP Voice amp Video Grandstream Networks Inc HT802 Analog Telephone Adaptor HT80X USER MANUAL http ArtTel ru This page is intentionally left bank http ArtTel ru dstream Innovative IP Voice amp Video HT802 USER MANUAL INDEX GNU GPL INFORMATION lt 4 WELGOME m 5 SAFETY COMPLAN O E G eure tette datu peau Dea es 5 22 5 CONNECT YOUNH MHIBUM i marania CURE QE 7 EQUIPMENT 400000 0000 0 7 CONNECTING THE 802 7 11 SOFTWARE FEATURES 11 HARDWARE 12 BASIC OPERATIONS bas Edi BEER aueh 13 UNDERSTANDING HT802 VOICE 13 PEAGING AA PHONE GALL tee 14 Phone or Extension 004 0 044 000000 14 IB euge EE 15 e BN IIOHP MC
41. nected to the FXS port Default value is 60 Sec Incoming call will stop ringing when not picked up given a specific period of time Default value is 20 seconds In case this feature activated using codes 92 code the call will be forwarded after this preconfigured amount of time Default is 4 seconds Dialing process is completed and outgoing call is initiated if no key entry occurs during this preconfigured interval Default is No Use only if proxy supports 484 response This parameter controls whether the phone will send an early INVITE each time a key is pressed when a user dials a number If set to Yes an INVITE is sent using the dial number collected thus far Otherwise no INVITE is sent until the Re Dial button is pressed or after about 5 seconds have elapsed if the user forgets to press the Re Dial button The Yes option should be used ONLY if there is a SIP proxy configured and the proxy server supports 484 Incomplete Address response Otherwise the call will likely be rejected by the proxy with a 404 Not Found error This feature does NOT work with and should NOT be enabled for direct IP to IP calling Sets the prefix added to each dialed number Default is Yes It allows users to configure the key as the Send or Dial key If set to Yes will send the number In this case this key is essentially equivalent to the Dial key If set to this 4 key be inclu
42. nection Request Username Connection Request Password CPE SSL Certificate CPE SSL Private Key System Ring Cadence Call Progress Tones Andstream Innovative IP Voice amp Video If set to Yes config file is authenticated before acceptance This protects the configuration from an unauthorized change The user specify SSL certificate used for SIP over TLS in X 509 format The user specify SSL private key used for SIP over TLS in X 509 format User specify password to protect the private key above User specify the Auto Configuration Server s URL TR 069 protocol User specify the ACS Username User specify the ACS password Default is No If set to YES device will send inform packets to the ACS Frequency that the inform packets will be sent out to the ACS Set a user name for the ACS to connect to this device Set a password for the ACS to connect to this device The Cert File for the phone to connect to the ACS via SSL The Cert Key for the phone to connect to the ACS via SSL Configuration option is set ring cadence on all FXS ports for all incoming calls Syntax c on1 off1 on2 off2 on3 off8 only cadences maximum Default is set to c 2000 4000 US standards Using these settings users can configure tone frequencies and cadence according to their preference By default they are set to North American frequencies Configure these settings with known values to avoid uncomfortable high pitch sounds ON is the period of
43. ng using cadence defined in parameter System Ring Cadence located under Advanced Settings Configuration page Note If server supports Alert Info header and standard ring tone set Bellcore or distinctive ring tone 1 10 is specified then the ring tone in the Alert Info header from server will be used Bellcore rings and tones are independent from custom ring tones The custom ring tones can also be specified by alert info header for example Alert Info lt http 127 0 0 1 gt info ring5 Default is No If set to YES Call Waiting indication information will not be provided to analog phone connected to this FXS port Default is No If set to YES Call Waiting caller ID will not be provided to analog phone connected to this FXS port Default is No This is to disable the stutter Call Waiting Tone when a Call Waiting information arrives The CWCID information will still be displayed Default is No If set to yes it will disable the warning to alert that the phone has been left off hook for an extended period of time Default is No Do not play the reminder ring when this is set to Yes Default is No If set to Yes the MWI information will not be transferred to the analog FIRMWARE VERSION 1 0 0 5 http ArtTel ru HT802 USER MANUAL Page 33 of 46 Ring Timeout Delayed Call Forward Wait Time No Key Entry Timeout Early Dial Dial Plan Prefix Use as Dial Key Andstream Innovative IP Voice amp Video phone con
44. one as the refresher or UAS to use the Callee or proxy server as the refresher Default is UAC As a Callee select UAC to use caller or proxy server as the refresher or UAS to use the phone as the refresher Session Timer can be refreshed using INVITE method or UPDATE method Select Yes to use INVITE method to refresh the session timer Default is No Default is No If set to Yes device will send an INVITE with audio vocoders upon completion of Fax to continue session in audio only Default is No If set to Yes Enables the use of PRACK Provisional Acknowledgment method Default is No If set to Yes device will include only the first match vocoder in its 200 response otherwise it will include all match vocoders in same order received in INVITE The HT802 supports up to 5 different Vocoder types including G 711 A U law G 726 32 G 723 1 G 729A B E iLBC The user can configure Vocoders in a preference list The first Vocoder is entered by choosing the appropriate option in Choice 1 The last Vocoder that will be included with the same preference order in SDP message is entered by choosing the appropriate option in Choice 6 Vocoder types can also be changed per call basis by using a star code Please see the Call features section Default is 2 Defines the number of voice frames sent in each packet Default is 6 3kbps Defines the encoding rate for G 723 1 vocoder Default is 20ms Sets the iLBC frame size in
45. re four 4 LED buttons that help you manage the status of your HandyTone 802 TABLE 2 BASIC DEFINITIONS OF THE HT802 LEDS PATTERN POWER LED Indicates Power Remains ON when power is connected Internet LED Indicates Access to Internet Remains ON while there is Access PHONE LED 1 amp 2 Indicate status of the respective FXS Ports PHONE on the back panel Unregistered OFF Registered and Available ON Solid Green Off Hook Busy Blinking every second Slow blinking FXS LEDs indicates voicemail NOTE All LEDs turns blue when ON Cordless Phone p LAN FXS p tt Analog A Fax Internet ADSL Cable HT802 Modem Ethernet FIGURE 2 HT802 CONNECTION DIAGRAM FIRMWARE VERSION 1 0 0 5 HT802 USER MANUAL Page 9 of 46 http ArtTel ru ustream Innovative IP Voice amp Video This page is intentionally left bank FIRMWARE VERSION 1 0 0 5 HT802 USER MANUAL Page 10 of 46 http ArtTel ru dstream Innovative IP Voice amp Video HT802 FEATRUES The HT802 is a full feature voice and fax over IP device that offers a high level of integration including a 10M 100Mbps network port and one FXS telephone port market leading sound quality rich functionalities and a compact and lightweight design The VoIP network signaling protocol supported is SIP The HT802 is fully compatible with SIP industry standard and can interoperate with many other SIP compliant devices and software on the market Moreover it suppo
46. ringing On time in ms while OFF is the period of silence In order to set a continuous tone OFF should be zero Otherwise it will ring ON ms and a pause of OFF ms and then repeat the pattern Example configuration for N A Dialtone 1 350 13 f2 440 13 c 0 0 Syntax f1 freq vol f2 freq vol c on1 off1 on2 off2 on3 off3 FIRMWARE VERSION 1 0 0 5 HT802 USER MANUAL Page 28 of 46 http ArtTel ru Andstream Innovative IP Voice amp Video Note freq 0 4000Hz vol 30 0dBm Feature Activation Blind Transfer Code Custom the Blind Transfer Code Default 90 Codes or followed by 1 5 digits Lock Keypad Update Default is No If set to Yes the configuration update via keypad is disabled Disable Voice Prompt Default is No Disables the voice prompt configuration Disable Direct IP Call Default is No Disables the Direct IP Call function URI or IP address of the NTP Network Time Protocol server This parameter NTP server synchronizes the date and time Allow DHCP option 42 Default Yes Enables the DHCP server to handle the NTP Server via Option 42 to override NTP serve Syslog Server The IP address or URL of System log server This feature is especially useful for the ITSP Internet Telephone Service Provider Syslog Level Select the HT802 to report the log level Default is NONE The level is one of EXTRA DEBUG DEBUG INFO WARNING or ERROR Syslog messages are sent based on the fo
47. rtTel ru ndstream Innovative IP Voice amp Video DHCP vendor class ID Default is HT7XX Used by clients and servers to exchange vendor specific information Time Zone Controls how the date time is displayed according to the specified time zone Self Defined Time Zone syntax is std offset dst offset start time end time Default is set to MTZ 6MDT 5 M3 2 0 M11 1 0 MTZ 6MDT 5 Time zone with 6 hours offset with 1 hour ahead which is the US central time It is positive if the local time zone is west of the Prime Meridian and negative if it is east Prime Meridian a k a International or Greenwich Meridian M3 2 0 M11 1 0 The 1 number indicates Month 1 2 3 12 for Jan Feb Dec The 274 number indicates the n iteration of the weekday 1st Sunday 3rd Tuesday etc The 3 number indicates Weekday 0 1 2 6 for Sun Mon Tue Sat Therefore this example is the DST which starts from the second Sunday of March to the 1st Sunday of November Allow DHCP server to Default Yes Let the DHCP server handle the Time Zone set Time Zone Language Languages supported with voice prompt and web interface except Spanish that it is only in IVR Reset Type Gives the user the option to set to default all VoIP related configuration mainly everything located on FXS Profile page all ISP Internet Service Provider configuration which may affect the IP address or both at the same time Note After you choos
48. rts comprehensive voice codecs including G 711 a p law G 723 1 G 726 32 G 729 and iLBC SOFTWARE FEATURES OVERVIEW e Supports Voice Codecs e G 711 a p law G 723 1 G 726 32 G 729 iLBC T 38 Fax e Comprehensive Dial Plan support for Outgoing calls 3 168 Echo Cancellation e Voice Activation Detection VAD Comfort Noise Generation CNG and Packet Loss Concealment PLC e Supports PSTN PBX analog telephone sets TABLE 3 HT802 SOFTWARE FEATURES Telephone Interfaces 2 FXS ports SIP Provisioning 2 Sip Accounts Number of 2 Concurrent Calls Concurrent Calls Voice over Packet Voice Activity Detection VAD with CNG comfort noise generation and PLC packet loss Capabilities concealment Dynamic Jitter Buffer Modem detection amp auto switch to G 711 Packetized Voice Protocol Unit supports RTP RTCP protocol G 168 compliant Echo Cancellation LEC line echo cancellation with NLP Asymmetric RTP stream Voice Compression G 711 Annex PLC Annex VAD CNG format encoder and decoder G 723 1 G 726 32 ADPCM G 729 iLBC G 726 32 provides proprietary VAD CNG and signal power estimation Voice Play Out unit reordering fixed and adaptive jitter buffer clock synchronization AGC automatic gain control Status output Decoder controlling via voice packet header FIRMWARE VERSION 1 0 0 5 HT802 USER MANUAL Page 11 of 46 http ArtTel ru DHCP Server Client Telnet Server Fax over Ip
49. sion will be terminated Session Expiration is the time in seconds at which the session is considered timed out if no successful session refresh transaction occurs beforehand The default value is 180 seconds The minimum session expiration in seconds The default value is 90 seconds Default is No If selecting Yes the phone will use session timer when it makes outbound calls if remote party supports session timer FIRMWARE VERSION 1 0 0 5 HT802 USER MANUAL Page 35 of 46 http ArtTel ru Callee Request Timer Force Timer UAC Specify Refresher UAS Specify Refresher Force INVITE Send Re INVITE After Fax Enable 100rel Use First Matching Vocoder in 2000K SDP Preferred Vocoder in listed order Voice Frames per TX G723 Rate iLBC Frame Size iLBC Payload type VAD Symmetric RTP Fax Mode Re Invite after Fax Tone Detection Mode Andstream Innovative IP Voice amp Video Default is No If selecting Yes the phone will use session timer when it receives inbound calls with session timer request Default is No If selecting Yes the phone will use session timer even if the remote party does not support this feature Selecting No will allow the phone to enable session timer only when the remote party support this feature To turn off Session Timer select No for Caller Request Timer Callee Request Timer and Force Timer Default is Omit As a Caller select UAC to use the ph
50. t SRTP squired RTP transmission mode Select between Disabled Enabled but not Forced and Enabled and Forced Default is Disabled Dependent on standard phone type and location Bellcore Telcordia ETSI FSK ETSI DTMF SIN 227 BT amp NTT Japan Brazil Default is No If set to Yes polarity will be reversed upon call establishment and termination Default is No Set it to Yes if the traditional PBX you are using with HT802 uses this method for signaling call termination Method initiates short voltage drop on the line when remote VoIP side disconnects an active call Default value is 200 Here can be configured duration of such voltage drop described in topic above HT802 supports a Duration Range from 100 to 10000 ms Time period when the cradle is pressed Hook Flash to simulate FLASH To prevent unwanted activation of the Flash Hold and automatic phone ring back adjust this time value Default values are 300 minimum and 1100 maximum HT802 supports a range from 40 to 2000 ms On hook timing is the minimum time for an on hook event to be validated Default value is 400 HT802 supports a range from 40 to 2000 ms Voice path volume adjustment e Rxis a gain level for signals transmitted by FXS e Txis a gain level for signals received by FXS Default OdB for both parameters Loudest volume 6dB Lowest volume 6dB User can adjust volume of call on either end using the Rx Gain Level parameter and the Tx Gain Leve
51. tone Then A can hang up NOTE Enable Call Feature must be set to Yes in web configuration page ATTENDED TRANSFER Assume that Caller A and B are in conversation Caller A wants to Attend Transfer B to C 1 Caller A presses FLASH on the analog phone for dial tone 2 Caller A then dials Caller C s number followed by or wait for 4 seconds 3 If Caller C answers the call Caller A and Caller C are in conversation Then A can hang up to complete transfer 4 f Caller C does not answer the call Caller can press flash to resume call with Caller B FIRMWARE VERSION 1 0 0 5 HT802 USER MANUAL Page 16 of 46 http ArtTel ru ustream Innovative IP Voice amp Video NOTE When Attended Transfer fails and A hangs up the HT802 will ring back user A to remind A that B is still on the call A can pick up the phone to resume conversation with B 3 WAY CONFERENCING The HT802 supports Bellcore style 3 way Conference Instructions for 3 way conference Assume that call party A and B are in conversation Caller A HT802 wants to bring third Caller C into conference 1 A presses FLASH on the analog phone or Hook Flash for old model phones to get a dial tone 2 Adials C s number then or wait for 4 seconds 3 If C answers the call then A presses FLASH to bring B C in the conference 4 If C does not answer the call A can press FLASH back to talk to B If A presses FLASH during conference C will b
52. u Device not registered This prompt will be played immediately after off hook If the device is not register and the option Outgoing Call without Registration is in NO Five Success Tips when using the Voice Prompt 1 shifts down to the next menu option 2 returns to the main menu 3 9 functions as the ENTER key in many cases to confirm or toggle an option 4 All entered digit sequences have known lengths 2 digits for menu option and 12 digits for IP address For IP address add 0 before the digits if the digits are less than 3 i e 192 168 0 26 should be key in like 192168000026 No decimal is needed 5 Key entry cannot be deleted but the phone may prompt error once it is detected PLACING A PHONE CALL PHONE OR EXTENSION NUMBERS 1 2 Dial the number directly and wait for 4 seconds Default No Key Entry Timeout or Dial the number directly and press Use as dial key must be configured in web configuration FIRMWARE VERSION 1 0 0 5 HT802 USER MANUAL Page 14 of 46 http ArtTel ru Innovative IP Voice amp Video Examples 1 Dial an extension directly on the same proxy e g 1008 and then press the or wait for 4 seconds 2 Dial an outside number e g 626 666 7890 first enter the prefix number usually 1 or international code followed by the phone number Press or wait for 4 seconds Check with your VoIP service provider for further details on pr
53. useful for ITSPs End user should keep it blank Firmware File Postfix Default is blank End user should keep it blank Config File Prefix Default is blank End user should keep it blank Config File Postfix Default is blank End user should keep it blank Allow DHCP Option 66 If set to Yes configuration and upgrade server information can be obtained using DHCP to override server option 66 from DHCP server located in customer s environment Default setting is Yes Automatic Upgrade Choose Yes to enable automatic upgrade and provisioning If select Check every minutes input the amount of minutes you want it to check for update If select Yes daily at hour make sure to input the hour of the day you want it to check for update e g for 11 pm type 23 If select Yes weekly on day make sure you input the day of the week in format 0 6 0 is Sunday you want it to check for update When set to No HT802 will only do the following option you select Always check for New Firmware at Boot up will check for new firmware every time the device reboots Check New Firmware only when F W pre suffix changes will check for updates only when the pre suffix has been changed FIRMWARE VERSION 1 0 0 5 HT802 USER MANUAL Page 27 of 46 http ArtTel ru Authenticate Conf File SIP TLS Certificate SIP TLS Private Key SIP TLS Private Key Password ACS URL ACS Username ACS Password Periodic Inform Enable Periodic Inform Interval Con
54. utes to reduce the Firmware or Provisioning Server load at any given time FIRMWARE VERSION 1 0 0 5 HT802 USER MANUAL Page 41 of 46 http ArtTel ru ustream Innovative IP Voice amp Video Automatic Upgrade C No C Yes every minutes 60 5256000 C Yes daily at hour 0 23 Yes weekly day 0 6 FIRMWARE VERSION 1 0 0 5 HT802 USER MANUAL Page 42 of 46 http ArtTel ru WARNING dstream Innovative IP Voice amp Video RESTORE FACTORY DEFAULT SETTING Restoring the Factory Default Setting will DELETE all configuration information of the phone Please BACKUP or PRINT out all the settings before you approach to following steps Grandstream will not take any responsibility if you lose all the parameters of setting and cannot connect to your VoIP service provider FACTORY RESET There are three 3 methods for resetting your unit RESET BUTTON Reset default factory settings following these four 4 steps 1 Unplug the Ethernet cable 2 Locate a needle sized hole on the back panel of the gateway unit next_to the power connection 3 Insert a pin in this hole and press for about 7 seconds 4 Take out the pin All unit settings are restored to factory settings IVR COMMAND Reset default factory settings using the IVR Prompt Table 5 1 Dial for voice prompt 2 Enter 99 and wait for reset voice prompt 3 Enter the encoded MAC address Look below on how to encode MAC
55. wed by Wait for dial tone then hang up 91 Cancel Busy Call Forward To cancel Busy Call Forward dial 91 wait for dial tone then hang up 92 Delayed Call Forward Dial 92 and then the forwarding number followed by Wait for dial tone then hang up 93 Cancel Delayed Call Forward To cancel Delayed Call Forward dial 93 wait for dial tone FIRMWARE VERSION 1 0 0 5 HT802 USER MANUAL Page 19 of 46 http ArtTel ru ustream Innovative IP Voice amp Video then hang up Flash Hook Toggles between active call and incoming call call waiting tone If not in conversation flash hook will switch to a new channel for a new call Pressing pound sign will serve as Re Dial key FIRMWARE VERSION 1 0 0 5 HT802 USER MANUAL Page 20 of 46 http ArtTel ru stream Innovative IP Voice amp Video CONFIGURATION GUIDE CONFIGURING THE HT802 THROUGH VOICE PROMPTS DHCP MODE Select voice menu option 01 to enable HT802 to use DHOP STATIC IP MODE Select voice menu option 01 to enable HT802 to use STATIC IP mode then use option 02 03 04 05 to set up IP address Subnet Mask Gateway and DNS server respectively FIRMWARE SERVER IP ADDRESS Select voice menu option 13 to configure the IP address of the firmware server CONFIGURATION SERVER IP ADDRESS Select voice menu option 14 to configure the IP address of the configuration server UPGRADE PROTOCOL Select voice menu option
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