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Grandstream Networks HandyTone 701 ATA

1. 13 HARDWARE SPECIFICATION 14 BASIC OPERA TIONS RE 17 UNDERSTANDING HT70X VOICE 17 PLACING A PHONE GALL centre lene det tiet eo na tee pete doeet cedet t bete eee epe ene 18 Phone or Extension Numbers esses sa assa assa 18 Direct IP 19 G7 E 0 Bl cre 20 20 CALL 20 VISTO e LL LLL M 20 Attended Transfer see eie ee tete eto eee eo 20 3 WAY 21 E TAES O EE 21 FIRMWARE VERSION 1 0 7 3 HT70X USER MANUAL Page 1 of 52 andstream Innovative IP Voice amp Video CALL FEATURES EM C UD FD E EAE 23 CONFIGURATION GUIDE 2 25 CONFIGURING THE HT70X THROUGH VOICE PROMPTS 25 CONFIGURING THE HT70X VIA WE
2. 49 RESTORE FACTORY DEFAULT SETTING 51 RE SE Leti iie e ertt veo tec eee be dte dedos Det cca bear ate ted dea 51 Reset IDONEOS ET 51 MR COMMA 51 Reset from web interface Reset 52 FIRMWARE VERSION 1 0 7 3 HT70X USER MANUAL Page 2 of 52 andstream Innovative IP Voice amp Video TABLE OF FIGURES HT70X USER MANUAL FIGURE 1 CONNECTING THE EITTOX tiere etie ite aeaaea vide aE A A a 10 FIGURE 2 HT70X CONNECTION emen nre 12 TABLE OF TABLES HT70X USER MANUAL TABLE 1 DEFINISIONS OF THE HT70X ener enne ANKANA RAKERA ANKA innere nnne nnn 10 TABLE 2 BASIC DEFINITIONS OF THE HT70X LEDS 10 TABLE 3 ADVANCED DEFINITIONS OF THE HT70X LEDS 11 TABLE 4 HT70X SOFTWARE FEATURES 2 13 TABLE 5 HT70X HARDWARE AND TECHNICAL SPECIFICATIONS 0 2 14 TABLE 6 HT70X IVR MENU DEFINITIONS 0 2 2 000000000000000000000 llli ninh ne 17 TABLE 7 HT70X CALL 2 23 TABLE 8 BASIC SETTINGS 27 TABLES STATUS PAGE
3. E EE 29 TABLE 0 ADVANCED SETTINGS i csse 30 TABLE 11 ACCOUNT SETTINGS 35 TABLE 12 HT704 FXS PORTS SETTINGS 44 CONFIGURATION GUI INTERFACE EXAMPLES HT70X USER MANUAL http www grandstream com products ht series ht701 documents ht70x gui zi 1 SCREENSHOT OF ADVANCED USER CONFIGURATION PAGE SCREENSHOT OF BASIC SETTINGS CONFIGURATION PAGE SCREENSHOT OF FXS PORT CONFIGURATION SCREENSHOT OF STATUS PAGE SCREENSHOT OF LOGIN PAGE SCREENSHOT OF REBOOT PAGE SCREENSHOT OF REBOOTING PAGE FIRMWARE VERSION 1 0 7 3 HT70X USER MANUAL Page 3 of 52 andstream Innovative IP Voice amp Video GNU GPL INFORMATION HT70X 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 VERSION 1 0 7 3 HT70X USER MANUAL Page 4 of 52 andstream Innovative IP Voice amp Video CHANGE LOG This section documents significant changes from previous version
4. Default is No If set to Yes device will challenge the incoming INVITE for the Authenticate ID and Password with 401 Unauthorized 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 If set to Default it will add Privacy header unless special feature is Telkom SA or CBCOM If set to Default it will add P Preferred Identity header unless special feature is Telkom SA or CBCOM Default is LAN Address If set to WAN Address device will detect its WAN address and use it in SIP REGISTER Contact Header 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 Configure the SIP Timer D defined in RFC3261 0 64 seconds Default 0 Sets the payload type for DTMF using RFC2833 Default is 101 The HT70X 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 RFC2833 events are generated until key is released Default is No If set to yes flash will be sent as DTMF ev
5. MAC Address Announces the Mac address of the unit 13 Firmware Server IP Announces current Firmware Server IP address Enter 12 digit new Address IP address 14 Configuration Server IP Announces current Config Server Path IP address Enter 12 digit Address new IP address 15 Upgrade Protocol Upgrade protocol for firmware and configuration update Press 9 to toggle between TFTP HTTP HTTPS FIRMWARE VERSION 1 0 7 3 HT70X USER MANUAL Page 17 of 52 andstream 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 CalP 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 menu 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
6. parameter will be attached to the From header in the SIP request to indicate the E 164 number If server supports TEL URI format then this option needs to be selected SIP Registration Controls whether the HT701 needs to send REGISTER messages to the proxy server The default setting is Yes FIRMWARE VERSION 1 0 7 3 HT70X USER MANUAL Page 36 of 52 Unregister on Reboot Outgoing Call without Registration Register Expiration Reregister before Expiration SIP Registration Failure Retry Wait Time Layer 3 QoS Local SIP port Local RTP port Use Random SIP Port Use Random RTP Port Refer to Use Target Contact Transfer on Conference Hang up Enable Ring Transfer Innovative IP Voice amp Video 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 HT70X 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 Default is 0 function disabled If configured with none 0 value Then this parameter will specify the time that HT70X refreshes its registration before expired instead of the default refresh time inte
7. 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 session 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 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 phone 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 Def
8. 222 FIRMWARE VERSION 1 0 7 3 HT70X USER MANUAL Page 51 of 52 andstream 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 000582006395 it should be keyed 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 HT70X 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 7 3 HT70X USER MANUAL Page 52 of 52
9. 250ms ON 250ms OFF 2x1000ms ON OFF T 3sec OFF 250ms ON 250ms OFF 1x250ms ON OFF T 3sec OFF 2x250ms ON OFF T 3sec OFF 3x250ms ON OFF T 3sec OFF FIRMWARE VERSION 1 0 7 3 HT70X USER MANUAL Page 11 of 52 andstream Innovative IP Voice amp Video LED 13 Receiver off hook test fail One or more phones are off 4x250ms ON OFF hook on phone line during test Phone 3sec OFF LED 14 REN test failed high REN detected Too many parallel 5x250ms ON OFF phones connected to phone line X Phone 3sec OFF LED 15 Line is active Phone 500ms ON OFF LED 16 Line inactive Phone ON LED 17 During Provisioning Stage Internet Pra 500ms ON OFF LED 18 During Firmware Recovery Internet 250ms ON OFF LED 19 Line X is registered normally to the sip providers network 2x1000 ms ON OFF and is ready to make a call Phone 3sec OFF Note In Provisioning and Firmware Recovery Stage the power LED is Steady ON Cordless Phone s LAN FXS gi Internet Analog ADSL Cable HT701 702 704 Modem Ethernet Fax FIGURE 2 HT70X CONNECTION DIAGRAM FIRMWARE VERSION 1 0 7 3 HT70X USER MANUAL Page 12 of 52 andstream Innovative IP Voice amp Video HT70X FEATRUES The HT70X 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 desig
10. 7 3 HT70X USER MANUAL Page 50 of 52 WARNING andstream 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 VolP 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 2 Unplug the Ethernet cable Locate a needle sized hole on the back panel of the gateway unit next to the power connection Insert a pin in this hole and press for about 7 seconds 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 address 4 Wait 15 seconds and device will automatically reboot and restore factory settings Encode the MAC Address 1 2 Locate the MAC address of the device It is the 12 digit HEX number on the bottom of the unit 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
11. 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 HT70X will stop the TFTP HTTP HTTPS process and simply boot using the existing code image in the flash 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 7 3 HT70X USER MANUAL Page 47 of 52 andstream 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 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 LOC
12. 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 7 3 HT70X USER MANUAL Page 24 of 52 andstream Innovative IP Voice amp Video CONFIGURATION GUIDE CONFIGURING THE HT70X THROUGH VOICE PROMPTS DHCP Select voice menu option 01 to enable HT70X to use DHCP STATIC IP MODE Select voice menu option 01 to enable HT70X 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 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 HT70X VIA WEB BROWSER HT70X 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 HT70X through a we
13. USER MANUAL Page 15 of 52 andstream Innovative IP Voice amp Video This page is intentionally left bank FIRMWARE VERSION 1 0 7 3 HT70X USER MANUAL Page 16 of 52 andstream Innovative IP Voice amp Video BASIC OPERATIONS UNDERSTANDING HT70X VOICE PROMPT HT70X 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 TABLE 6 HT70X IVR MENU DEFINITIONS OT PTIONS Main Menu Enter a Menu Option 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 01 DHCP Mode Press 9 to toggle the selection Static IP Mode If using Static IP Mode configure the IP address information using menus 02 to 05 If using Dynamic IP Mode all IP address information comes from the DHCP server automatically after reboot 02 IP Address address 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 03 Subnet IP address Same as menu 02 04 Gateway IP address Same as menu 02 05 DNS Server IP address Same as menu 02 07 Preferred Vocoder Press 9 to move to the next selection in the list PCMU PCMA e G 726 e 80 723 e G 729 10
14. amp Video e xx atleast 2 digits number e xx atleast 2 digit number e 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 1 gt add a leading 1 to all numbers dialed vice versa will remove a 1 from the number dialed or Example 1 369 11 1617 Allow 311 611 911 and any 11 digit numbers with leading digits 1617 Example 2 1900x lt 1617 gt xxxxxxx Block any number of leading digits 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 reading 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 011 3469 11
15. hang up to complete transfer 4 If Caller C does not answer the call Caller A can press flash to resume call with Caller B FIRMWARE VERSION 1 0 7 3 HT70X USER MANUAL Page 20 of 52 andstream Innovative IP Voice amp Video NOTE When Attended Transfer fails and A hangs up the HT70X 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 HT701 702 704 supports Bellcore style 3 way Conference Instructions for 3 way conference Assume that call party A and B are in conversation Caller A HT70X 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 A dials C s number then or wait for 4 seconds If C answers the call then A presses FLASH to bring B C in the conference If C does not answer the call A can press FLASH back to talk to B If A presses FLASH during conference C will be dropped out o 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 HT70X 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 con
16. 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 tone Then A can hang up NOTE Enable Call Feature must be set to Yes 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
17. 00 Mbps 12V51A orts with LEDS FIGURE 1 DIAGRAM OF HT70X TABLE 1 DEFINISIONS OF THE HT70X CONNECTORS DC 12V Power adapter connection Internet Port RJ 45 Connect to the internal LAN network HT701 Only LAN Port RJ 45 Connect to the internal LAN network HT702 and HT704 Only RESET Factory Reset button Press for 7 seconds to reset factory default settings Phone Port s RJ 11 FXS port to be connected to analog phones fax machines There are four 4 LED buttons that help you manage the status of your HandyTone 701 and there are five 5 LED buttons that help you manage the status of your HandyTone 702 and 704 TABLE 2 BASIC DEFINITIONS OF THE HT70X LEDS PATTERN POWER LED Indicates Power Remains ON when power is connected Internet LED Indicates Access to Internet Remains ON while there is Access HT701 and HT702 Only Link Activity LED Indicates if There is Activity on the Internet Port HT701 and HT702 Only PHONE LED Indicate status of the respective FXS Ports PHONE on the back panel FIRMWARE VERSION 1 0 7 3 HT70X USER MANUAL Page 10 of 52 Unregistered OFF Registered and Available ON Solid Green Off Hook Busy Blinking every second Slow blinking FXS LEDs indicates voicemail NOTE All LEDs display green when ON LED 01 LED 02 LED 03 LED 04 LED 05 LED 06 LED 07 LED 08 LED 09 LED 10 LED 11 LED 12 TABLE 3 ADVANCED DEFINITIONS OF THE HT70X LEDS PATTERN Device h
18. AL 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 HT70X web interface Advanced Settings page you can browse your hard drive and select the folder you previously saved the file ht7Oxfw 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 HT701 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 Startthe 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 Grandstream 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 7 3 HT70X USER MANUAL Page 48 of 52 andstream Innovative IP Voice amp Vide
19. B BROWSER eene nemen 25 Access the Web Configuration sns nnns nnns 25 IMPORTANT 5 85 2 1 4404 1 4 1 1 a a a a Aaa aE a A anaa 26 NAT Sna ER 26 DIM MeO S ent e ap n Dose e ee aN ea buius 27 Preferred VOCODER 27 SAVING THE CONFIGURATION nennen 45 REBOOTING THE HT70X FROM 46 CONFIGURATION THROUGH A CENTRAL SERVER 2 200 4 2 0 04 50 0 46 SOFTWARE UPGRADE 47 FIRMWARE UPGRADE THROUGH 47 INSTRUCTIONS FOR UPLOAD FROM LOCAL 48 INSTRUCTIONS FOR LOCAL FIRMWARE UPGRADE USING TFTP SERVER 48 CONFIGURATION FILE DOWNLOAD ssseen eene nnnm enne nennen nennen nennen nn 48 FIRMWARE AND CONFIGURATION FILE PREFIX AND POSTFIX 49 MANAGING FIRMWARE AND CONFIGURATION FILE
20. DE 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 Firmware upgrade server in 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 Grandstream recommends end user use the Grandstream HTTP server Its address can be found at http www grandstream com support firmware Currently the HTTP firmware server address is firmware grandstream com For large companies we recommend to maintain their own TFTP HTTP HTTPS server for upgrade and provisioning procedures 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 HT70X will attempt to retrieve the new image files by downloading them into the HT70X s SRAM During this stage the HT70X 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
21. NT PACKAGING The HT70X ATA package contains e One HT70X Main Case e One Universal Power Adaptor e One Ethernet Cable e One Vertical Stand Only on HT702 and HT704 Packages CONNECTING THE HT70X The HT70X is designed for easy configuration and easy installation Configure the HT70X following the 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 HT70X 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 HT70X and connect it to a wall outlet The HT70X Analog Telephone Adaptor is an all in one VoIP integrated device designed to be a total solution for networks providing VoIP services The HT70X VoIP features and functions are available using a regular analog telephone FIRMWARE VERSION 1 0 7 3 HT70X USER MANUAL Page 9 of 52 andstream Innovative IP Voice amp Video HT 701 RJ 11 FXS Ports Internet Port Power RJ 45 connector Display LEDs Suppl 10 100 Mbps Reset green ps 12V 0 5A HT702 HT704 Display LED s green Display LED s green LAN Port Power Phone RJ 11 LAN Port Power RJ 45 connector Reset Supply FXS Ports RJ 45 connector Reset Supply 10 100 Mbps 12V 1A 10 1
22. SLIC chip exception WARNING and ERROR levels Oo mi Qu sg G5 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 is up Default is No If Enabled the HT70X will send a replicate of the SIP packets on the Send SIP Log css Set the IP or FQDN of the Primary RADIUS Server HT704 Only Remote Authentication Dial In User Service RADIUS HT704 supports RADIUS for authentication authorization and billing purposes Primary and secondary RADIUS server configurations are available to provide redundancy to this feature In case Primary Radius Primary RADIUS S server becomes unusable RADIUS requests will be automatically sent to the secondary erver server When at least one RADUIS server was configured the device will allow users to make phone calls only after authorization from RADIUS server has been received CDR Call Detail Record is also sent to the RADIUS server for billing purposes RAIDUS server can send requests to terminate calls when run out of pre paid credit FIRMWARE VERSION 1 0 7 3 HT70X USER MANUAL Page 34 of 52 Primary RADIUS Auth Port Primari RADIUS Acct Port Primary RADIUS Server Secret Se
23. Sat Therefore this example is the DST which starts from the second Sunday of March to the 1st Sunday of November FIRMWARE VERSION 1 0 7 3 HT70X USER MANUAL Page 28 of 52 Allow DHCP server to set Time Zone Language Reset Type andstream Innovative IP Voice amp Video Default Yes Let the DHCP server handle the Time Zone Languages supported with voice prompt and web interface except Spanish that it is only in IVR 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 choose 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 MAC Address IP Address Product Model Hardware Version Software Version System Up Time PPPoE Link Up NAT TABLE 9 STATUS PAGE 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 This field shows the IP address of the HT70X This field contains the product model info This field shows the hardware revision of the
24. This setting should be based on your server DTMF setting PREFERRED VOCODER CODEC The HT70X supports a broad range of voice codecs Under Profile web pages choose your preferred order of different codecs or G711y a e G729 G723 1 e 3726 16 24 32 40 e iLBC TABLE 8 BASIC SETTINGS End User Password Password to access the Web Configuration Menu This field is case sensitive with a maximum length of 25 characters Web Port By default HTTP uses port 80 This field is for customizable web port Telnet Server Default is set to Yes HTTP Access Default is set to Yes If set to No http access will be denied FIRMWARE VERSION 1 0 7 3 HT70X USER MANUAL Page 27 of 52 IP Address DHCP hostname DHCP domain DHCP vendor class ID PPPoE account ID PPPoE password PPPoE Service Name Preferred DNS server Time Zone Self Defined Time Zone Andstream Innovative IP Voice amp Video There are two modes to operate the HT70X DHCP mode all the field values for the Static IP mode are not used even though they are still saved in the Flash memory The HT701 acquires its IP address from the first DHCP server it discovers from the LAN it is connected Using the PPPoE feature set the PPPoE account settings The HT70x will establish a PPPoE session if any of the PPPoE fields is set Static IP mode configure the IP address Subnet Mask Default Router IP address ie Preferred DNS Server 219 Preferre
25. X USER MANUAL Page 6 of 52 andstream Innovative IP Voice amp Video WELCOME Thank you for purchasing Grandstream s HT70X the affordable feature rich Analog Telephone Adaptor Grandstream HandyTone70X 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 HT70X is very easy to manage and configure and is specifically designed to be an easy to use and affordable VolP solution for both the residential user and the teleworker SAFETY COMPLIANCES The HT70X phone complies with FCC CE and various safety standards The HT70X power adaptor is compliant with UL standard Only use the universal power adapter provided with the HT70X package The manufacturers warranty does not cover damages to the phone caused by unsupported power adaptors WARRANTY If you purchased your HT70X 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 Service Representative for a RMA Ret
26. 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 Subscribe for MWI Default is No When set to Yes a SUBSCRIBE for Message Waiting Indication will be sent periodically Send Anonymous 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 FIRMWARE VERSION 1 0 7 3 HT70X USER MANUAL Page 41 of 52 Anonymous Call Rejection Special Feature Session Expiration Min SE Caller Request Timer Callee Request Timer Force Timer UAC Specify Refresher UAS Specify Refresher Force INVITE Send Re INVITE After Fax Enable Silence Detection for Fax Disconnect Enable 100rel Add Auth Header On Initial REGISTER Use First Matching Vocoder in 2000K SDP Preferred Vocoder Andstream Innovative IP Voice amp Video 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
27. an transfer the call upon receiving ring back tone or SIP message 180 FIRMWARE VERSION 1 0 7 3 HT70X USER MANUAL Page 37 of 52 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 Authenticate incoming INVITE Allow Incoming SIP Messages from SIP Proxy Only Use Privacy Header Use P Preferred Identity Header SIP REGISTER Contact Header Uses SIP T1 Timeout SIP T2 Interval SIP Timer D DTMF Payload Type Preferred DTMF method Disable DTMF Negotiation Generate Continuous RFC2833 Events Send Flash Event Enable Call Features Andstream Innovative IP Voice amp Video 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 defined 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
28. as normal power Power Error Power is removed from the device or power supply with improper voltage is plugged in Line X is registered normally to the sip providers network and is ready to make call Voice mail waiting for Line X Device has normal WAN connection and has obtained IP address Internet link error Device is powered up and ready to connect to the Internet but the WAN INTERNET port is down Internet DHCP Error Device is properly connected but it is unable to retrieve an IP address from the device it is connected to Line Registration failed Device is properly setup can connect to provider s network but cannot register to provider s SIP proxy no 200 OK Device is connected has physical data link but there are incorrect network settings typically associated with PPPoE connection failure Hazardous potential test failed Hazardous AC or DC voltage is present on the tip and ring or both signals of phone line X Foreign electro Motive Force EMF Test fail Foreign voltage is present on the tip ring or both signals of phone line Device has detected additional external Phone voltage on the FXS phone line Resistive fault test failed Either tip or ring is shorted to ground or they are shorted to each other Power Power Phone Phone Internet Internet Internet Phone Internet Phone Phone Phone andstream Innovative IP Voice amp Video ON OFF ON 1sec ON 3sec OFF ON OFF
29. ault 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 For fax machines that do not send a Disconnect when fax is done This option Enables Disables the detection of silence in order to know the fax has finished The silence period is non configurable and fixed to 7 seconds Default is No Default is No If set to Yes Enables the use of PRACK Provisional Acknowledgment method If set to Yes device will include authorization header in the Register request 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 HT70X 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 that will be included with the same preference order in SDP message The first FIRMWARE VERSION 1 0 7 3 HT70X USER MANUAL Page 42 of 52 Voice Frames per TX G723 Rate iLBC Frame Size iLBC Payload type VAD Symmetric RTP Fax Mode Re Invite after Fax Tone Detection Mode Jitter Buffer Type Jitter Buffer Length SRTP Mode SLIC Setting Caller ID Scheme DTMF Caller ID Tone Polarity Reversal Loop Current Disconnect Loop Current Disconnect Duration Enable Hook Flash Hook Flash Timing Andstream Inn
30. b 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 HT70X using voice prompt menu option 02 FIRMWARE VERSION 1 0 7 3 HT70X USER MANUAL Page 25 of 52 andstream 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 a 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 admin 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 5 configuration pages Please reference the GUI pages using the following link http www grandstream com products ht series ht701 documents ht70x gui zip NOTE If you cannot log into the configuration page by using the default password please check wit
31. condary RADIUS Server Secondary RADIUS Auth Port Secondary RADIUS Acct Port Secondary RADIUS Server Secret RADIUS Timeout RADIUS Retry Download Device Configuration Upload Firmware Profile Account Active Primary SIP Server Failover SIP Server Prefer Primary SIP Server Innovative IP Voice amp Video Default is 1812 Specifies the port to be used for the Primary RADIUS Authentication HT704 Only Default is 1813 Specifies the port to be used for the Primary RADIUS Account HT704 Only Specifies the secret string to be used to authenticate the RADIUS connection to the Primary Server It should match RADIUS configuration HT704 Only Set the IP or FQDN of the Secondary RADIUS Server HT704 Only In case Primary Radius server becomes unusable secondary will take role of primary and Will manage credit recourses in the network Default is 1812 Specifies the port to be used for the Secondary RADIUS Authentication HT704 Only Default is 1813 Specifies the port to be used for the Secondary RADIUS Account HT704 Only Specifies the secret string to be used to authenticate the RADIUS connection to the Secondary Server It should match RADIUS configuration HT704 Only Default is 2 HT704 Only Default is 3 HT704 Only 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 Download
32. configuration completed in Distinctive Ring Tones block in the same page applies to ring tones cadences configured here TABLE 12 HT704 FXS PORTS SETTINGS 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 subscriber s Authenticate ID used for authentication Can be identical to or different from SIP User ID SIP service subscriber s account password for HT704 to register to SIP servers of ITSP FIRMWARE VERSION 1 0 7 3 HT70X USER MANUAL Page 44 of 52 Name Profile ID Hunting Group Enable Ports Offhook Auto dial Map to FXP Port Map to FXO Gateway IP and Port Andstream Innovative IP Voice amp Video Any name to identify this specific user Select the corresponding Profile ID between Profile 1 and Profile 2 This feature enables the HT704 to register all existing FXS ports with the same phone number Each incoming call will be routed to first available port in Linear or Circular mode User may configure all ports as members of the same Hunting Group or it may configure different port combinations for more than one Hunting Group For example Ports 1 3 and 5 are members of the same Hunting Group the rest of the ports may have separate numbers and may be reached independently Any port member of a Hunting Group that is not registered with a SIP account will be able to place outbound calls usi
33. d 1100 maximum HT70X 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 HT70X supports a range from 40 to 2000 ms Voice path volume adjustment e Rxisa gain level for signals transmitted by FXS 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 Level 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 No If set to Yes NEC will be disabled per call base Recommended for FAX Data calls Configure the time limitation of outgoing calls 0 180 minutes default is 0 No Limit Configure ringing frequency for your phone 15 60 Hz default is 20 Hz Default is No If it has difficulties to ring the FAX machine or phone Enable to push higher ring power through FXS port This function lets you configure ring tone cadence preferences User has 10 choices The
34. d DNS Server 3 Preferred DNS Server 4 Preferred DNS Server These fields are set to zero by default Default is blank This option specifies the name of the client This field is optional but may be required by some Internet Service Providers Default is blank This option specifies the domain name that client should use when resolving hostnames via the Domain Name System Default is HT7XX Used by clients and servers to exchange vendor specific information PPPoE username Necessary if ISP requires you to use a PPPoE Point to Point Protocol over Ethernet connection PPPoE account password Default is blank This field is optional If your ISP uses a service name for the PPPoE connection enter the service name here The preferred DNS Server to be used Controls how the date time is displayed according to the specified time zone The 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 15 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
35. ditions 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 7 3 HT70X USER MANUAL Page 21 of 52 andstream Innovative IP Voice amp Video This page is intentionally left bank FIRMWARE VERSION 1 0 7 3 HT70X USER MANUAL Page 22 of 52 Andstream Innovative IP Voice amp Video CALL FEATURES The HT70X supports all the traditional and advanced telephony features 02 03 16 17 30 31 47 50 51 67 82 69 70 71 72 73 74 78 79 87 90 91 92 93 TABLE 7 HT70X CALL FEATURES CALL FEATURES Forcing a Codec per call 027110 PCMU 027111 PCMA 02723 G723 02729 G729 0272616 G726 r16 0272624 G724 r24 0272632 0726 32 0272640 G726 r40 027201 iLBC Disable LEC per call Dial 03 number No dial tone is played in the middle Enable SRTP Disable SRTP Block Caller ID for all subsequent calls Send Caller ID for all subsequent calls Direct IP Calling Dial 47 IP address dial tone is played in the middle Detail see Direct IP Calling section on page 12 Disable Call Waiting for all subsequent calls Enable Call Waiting for all subsequent calls Block Caller ID per call Dial 67 number No dial ton
36. e is played in the middle Send Caller ID per call Dial 82 number No dial tone is played in the middle Call Return Service Dial 69 and the phone will dial the last incoming phone number received Disable Call Waiting per call Dial 70 number No dial tone is played in the middle Enable Call Waiting per call Dial 71 number No dial tone is played in the middle Unconditional Call Forward Dial 72 and then the forwarding number followed by Wait for dial tone and hang up dial tone indicates successful forward Cancel Unconditional Call Forward To cancel Unconditional Call Forward dial 73 wait for dial tone then hang up Enable Paging Call Dial 74 and then the destination phone number you want to page Enable Do Not Disturb DND When enabled all incoming calls are rejected Disable Do Not Disturb DND When disabled incoming calls are accepted Blind Transfer Busy Call Forward Dial 90 and then the forwarding number followed by Wait for dial tone then hang up Cancel Busy Call Forward To cancel Busy Call Forward dial 91 wait for dial tone then hang up Delayed Call Forward Dial 92 and then the forwarding number followed by Wait for dial tone then hang up Cancel Delayed Call Forward To cancel Delayed Call Forward dial 93 wait for dial tone FIRMWARE VERSION 1 0 7 3 HT70X USER MANUAL Page 23 of 52 andstream
37. ect IP Call Default is No Disables the Direct IP Call function Failover to FXO Default is Disable If Enabled the unit will start routing the calls to the FXO Gateway Gateway configured via Direct IP Call When it loses registration from your SIP Server HT704 Only The use of this option presumes a configured GXW410x or another FXO gateway with an active PSTN line connection FXO Gateway IP or Domain Name of the FXO Gateway that will be used as Failover HT704 Only URI or IP address of the NTP Network Time Protocol server This parameter NTP server synchronizes the date and time FIRMWARE VERSION 1 0 7 3 HT70X USER MANUAL Page 33 of 52 andstream Innovative IP Voice amp Video 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 HT701 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 following 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
38. ed Allows the user to upgrade the firmware with a single firmware file by browsing and loading the file from your computer local directory TABLE 11 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 FIRMWARE VERSION 1 0 7 3 HT70X USER MANUAL Page 35 of 52 Innovative IP Voice amp Video Outbound Proxy IP address or Domain name of Outbound Proxy or Media Gateway or Session Border Controller Used by HT70X 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 Allow DHCP Option Default is No If set to Yes device will collect SIP server address from DHCP option 120 override SIP server 120 for FXS1 HT701 2 or 1 HT704 SIP transport User can select UDP or TCP or TLS Default is UDP NAT Traversal STUN This parameter defines whether or not the HT70X NAT traversal mechanism is activated If activated by choosing Yes and a STUN server is also specified then the HT70X performs according to the STUN client specification Using this mode the embedded STUN client will detect if and what t
39. ent Default is Yes If Yes call features using star codes will be supported locally FIRMWARE VERSION 1 0 7 3 HT70X USER MANUAL Page 38 of 52 Offhook Auto Dial Offhook Auto Dial Delay Proxy Require Use NAT IP Use SIP User Agent Header Do Not Escape as 9623 in SIP URI Disable Multiple m line in SDP Distinctive Ring Tone Disable Call Waiting Disable Call Waiting Caller ID Disable Call Waiting Tone Disable Receiver 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 HT70X will automatically append the and the host portion of the corresponding SIP address HT701 and HT702 only 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 If set to Yes device will use instead of 23 in the send URI Default is No If set to Yes device will send only one m line in SDP regardless how many m field in the incoming SDP 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 Ca
40. ent to Sip port that will be annexed to the IP address above SAVING THE CONFIGURATION CHANGES Click the Update button in the Configuration page to save the changes to the HT70X configuration The following screen confirms that the changes are saved Reboot or power cycle the HT70X to make the changes take effect FIRMWARE VERSION 1 0 7 3 HT70X USER MANUAL Page 45 of 52 andstream Innovative IP Voice amp Video REBOOTING THE HT70X 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 CONFIGURATION THROUGH A CENTRAL SERVER Grandstream HT70X can be automatically configured from a central provisioning system When HT70X boots up it will send TFTP or HTTP HTTPS requests to download configuration files cfg000b82xxxxxx and 000082 where 000b82xxxxxx is the LAN MAC address of the HT70X If the download of cfgxxxxxxxxxxxx xml is not successful the provision program will issue request a generic configuration file cfg xml Configuration file name should be in lower case letters The configuration data can be downloaded via TFTP or HTTP HTTPS from the central server A service provider or an enterprise with large deployment of HT70X can easily manage the configuration and service provisioning of ind
41. ffhook Tone Disable Reminder Ring for On Hold Call Disable Visual MWI Ring Timeout Hunting Group Ring Timeout Hunting Group Type Delayed Call Forward Wait Time No Key Entry Timeout Early Dial Dial Plan Prefix Use as Dial Key Dial Plan Andstream Innovative IP Voice amp Video 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 phone connected 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 Sec If call is not answered within this designated time period the callwill be forwarded to the next member of a Hunt Group HT704 only Linear and Circular Default is Circular Linear style will sort the call to the lowest numbered available line this is also called serial hunting Circular style will distribute the calls round robin If a call is assigned to line 1 the next call goes to 2 and the next to 3 The succession throughout each of the lines continues even if one of the previous lines becomes available When the end of the hunt group is reached the hunting starts over at the firstline Lines are skipped if they are still busy on a previous call HT704 only Default value is 20 seconds In case this feature activated using codes 92 code the call wil
42. h the VoIP service provider It is most likely the VoIP service provider has provisioned the device and configured for you therefore the password has already been changed 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 FIRMWARE VERSION 1 0 7 3 HT70X USER MANUAL Page 26 of 52 andstream Innovative IP Voice amp Video 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 Set this to Yes when gateway is behind firewall on a private network DTMF METHODS DTMF Settings are in FXS portX ProfileX page e DTMF in audio e DTMF via RTP RFC2833 DTMF via SIP INFO Set priority of DTMF methods according to your preference
43. isioning 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 HT70X 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 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 FIRMWARE VERSION 1 0 7 3 HT70X USER MANUAL Page 32 of 52 Andstream Innovative IP Voice amp Video Connection Request Set a user name for the ACS to connect to this device Username Connection Request Set a pas
44. ividual devices remotely from a central server Grandstream provides a central provisioning system GAPS Grandstream Automated Provisioning System to support automated configuration of Grandstream devices GAPS uses enhanced NAT friendly TFTP or HTTP thus no NAT issues and other communication protocols to communicate with each individual Grandstream device for firmware upgrade remote reboot etc Grandstream provides GAPS service to VoIP service providers Use GAPS for either simple redirection or with certain special provisioning settings At boot up Grandstream devices by default point to Grandstream provisioning server GAPS based on the unique MAC address of each device GAPS provision the devices with redirection settings so that they will be redirected to customer s TFTP or HTTP HTTPS server for further provisioning Grandstream also provides configuration tools Windows and Linux Unix version to facilitate the task of generating device configuration files The Grandstream configuration tools are free to end users The configuration tools configuration templates are available for download from http Awww grandstream com support tools FIRMWARE VERSION 1 0 7 3 HT70X USER MANUAL Page 46 of 52 andstream 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 UPGRA
45. jn dsiream Innovative IP Voice amp Video Grandstream Networks Inc HT701 HT702 HT704 Analog Telephone Adaptor HT701 HT702 HT704 HT70X USER MANUAL This page is intentionally left bank andstream Innovative IP Voice amp Video HT70X USER MANUAL INDEX GNU GPL INFORMATION 4 CHANGE LOG Y 5 CHANGES FROM 1 0 6 1 USER 5 CHANGES FROM 1 0 5 2 USER MANUAL 4 5 CHANGES FROM 1 0 4 14 USER 5 CHANGES FROM 1 0 4 8 USER 5 CHANGES FROM 1 0 3 1 USER 6 CHANGES FROM 1 0 1 6 USER 6 CHANGES FROM 1 0 0 18 USER 6 EN 7 SAFETY COMPLIANCES 0 0 ER CUN 7 5 cot ee oes 7 CONNECT 5 9 EQUIPMENT 040 0000000 nnn nanan unAn siiis s seda sane ns s sss sans 9 CONNECTING THE 9 HIDE FEATRUES eee eae Ma SEM ERE 13 SOFTWARE FEATURES
46. l 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 No this key be included as part of number 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 FIRMWARE VERSION 1 0 7 3 HT70X USER MANUAL Page 40 of 52 andstream Innovative IP Voice
47. le 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 minutes to reduce the Firmware or Provisioning Server load at any given time FIRMWARE VERSION 1 0 7 3 HT70X USER MANUAL Page 49 of 52 andstream Innovative IP Voice amp Video Automatic Upgrade e No Yes m minutes 60 5256000 e Yes daily at 0 23 e Yes weekly on day 0 6 FIRMWARE VERSION 1 0
48. ller 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 ring 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 FIRMWARE VERSION 1 0 7 3 HT70X USER MANUAL Page 39 of 52 O
49. n The VoIP network signaling protocol supported is SIP The HT70X is fully compatible with SIP industry standard and can interoperate with many other SIP compliant devices and software on the market Moreover it supports 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 G 711 a p law G 723 1 G 726 32 G 729 and iLBC e T 38 Fax e Comprehensive Dial Plan support for Outgoing calls e 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 4 HT70X SOFTWARE FEATURES HT 701 HT 702 HT 704 Telephone Interfaces 1 FXS ports 2 FXS ports 4 FXS ports SIP Provisioning 1 Sip Account 1 Profile 2 Sip Accounts 2 Profiles 4 Sip Accounts 2 Profiles Number of 1 Concurrent Call 2 Concurrent Calls 4 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 9 711 Annex PLC Annex Il VAD CNG format encoder and decoder G 723 1 G 726 32 ADPCM G 729 iLBC G 726 32 provides proprietar
50. nSSL This is required for the phone to decrypt the encrypted XML configuration file The user name needed to authenticate with the HTTP HTTPS server The password needed to authenticate with the HTTP HTTPS server FIRMWARE VERSION 1 0 7 3 HT70X USER MANUAL Page 31 of 52 Always send HTTP Basic Authentication Information Firmware File Prefix Firmware File Postfix Config File Prefix Config File Postfix Allow DHCP Option 66 to override server Automatic Upgrade 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 andstream Innovative IP Voice amp Video Default is Send HTTP Basic Authentication Information only when challenged If set to Always device will send configured user name and password within HTTP request without server sending authentication challenge Default is blank If configured HT701 will request firmware file with the prefix This setting is useful for ITSPs End user should keep it blank Default is blank End user should keep it blank Default is blank End user should keep it blank Default is blank End user should keep it blank If set to Yes configuration and upgrade server information can be obtained using DHCP option 66 from DHCP server located in customer s environment Default setting is Yes Choose Yes to enable automatic upgrade and prov
51. ng the SIP credentials of the primary Hunting Group port For example Port 1 2 and 3 are members of the same Hunting Group Port 1 is registered with a SIP account Ports 2 and 3 are not registered Ports 2 and 3 will be able to place outbound calls using the SIP account of port 1 Select appropriate value for Hunting Group feature The original SIP account should be set to Active while the group members should be set to the port number of the Active Port Example configuration of a Hunting group FXS Port 1 SIP UserlD and Authenticate ID entered Hunting group set to Active FXS Port 42 SIP UserlD and Authenticate ID left blank Hunting Group set to 1 FXS Port 3 SIP UserlD and Authenticate ID left blank Hunting Group set to 1 FXS Port 4 SIP UserlD and Authenticate ID entered Hunting group set to None Hunting Group 1 contains ports 1 2 3 FXS port 4 is registered but it is not added to the Hunting Group 1 Set No to disable FXS port This feature allows you to automatically dial the number specified in this field as soon as the port is offhooked i e when the receiver on the phone connected to Port is picked up This is used only when peering with a Grandstream GXW410x Default is 1 Supported values 1 8 meaning line 1 to line 8 of the GXW410x device where the port will be mapped to This is used when peering with an FXO gateway of any brand You have to specifically mention the IP address where the call will be s
52. nsfer 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 HT70X Telephone 1 RJ11 FXS port Interfaces Network Interface LED Indicators 1 RJ45 10 100 Mb Base TX Full Duplex POWER INTERNET TABLE 5 HT70X HARDWARE AND TECHNICAL SPECIFICATIONS HT701 HT702 HT704 2 RJ11 FXS ports 4 RJ11 FXS ports 1 RJ45 10 100 Mb Base TX Full Duplex port with connectivity LEDs POWER INTERNET POWER PHONE1 PHONE2 FIRMWARE VERSION 1 0 7 3 HT70X USER MANUAL Page 14 of 52 Factory Reset Button Universal Switching Power Adaptor Environmental Dimensions H x W x D Short Haul Loop Polarity Reversal Wink EMC Safety Compliance andstream Innovative IP Voice amp Video LINK ACTIVITY PHONE LINK ACTIVITY PHONE3 PHONE4 PHONE1 PHONE2 Yes Input 100 240 VAC 50 60 Hz 0 18A Max Output 12VDC 1A UL certified Input 100 240 VAC 50 60 Hz 0 18A Max Ouiput 12VDC 0 5A UL certified Operational 32 104 F or 0 40 C Storage 14 140 F or 10 60 Humidity 10 90 Non condensing 26 x 65 x 86mm 28x115 x 75mm 28x115 x 75mm 5REN Up to 1Km on 24 AWG 3REN Up to 1Km on 24 AWG line wire Yes FCC part15 Class B EN55022 EN55024 CISPR22 and CISPR24 EN60950 1 amp UL60950 1 UL only for PSU Fe FIRMWARE VERSION 1 0 7 3 HT70X
53. o 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 page 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 C JXXXXXXXXXXXX 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 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 sing
54. o enable disable Allow DHCP Option 120 override SIP server e Added the option DTMF Caller ID Tone to define DTMF call ID Start Stop Tone FIRMWARE VERSION 1 0 7 3 HT70X USER MANUAL Page 5 of 52 andstream Innovative IP Voice amp Video CHANGES FROM 1 0 3 1 USER MANUAL e Added the option to enable disable HTTP Access HTTP Access e Added the option to enable disable Authenticate incoming INVITE Authenticate incoming INVITEHTTP Access e Added ability to configure the time of re register before registration expired Reregister before Expiration e Updated Table3 Advanced Definitions of the HT70X LEDs Pattern e Added the option to enable disable Use DNS to detect network connectivity Use DNS to detect network connectivity CHANGES FROM 1 0 1 6 USER MANUAL e Added the option to enable disable hook flash Enable Hook Flash e Removed firmware key from Advanced Setting page CHANGES FROM 1 0 0 18 USER MANUAL e Added ability to configure delay for the off hook auto dial Offhook Auto Dial Delay e Added display of gs version in status page CPE e Added a configuration parameter to overdrive User Agent header Use SIP User Agent Header e Added CPE SSL Certificate and CPE SSL Private Key in Advanced web page e Added an option to Enable Disable each FXS Port Enable Ports e Split function Use Random Port into Use Random SIP Port and Use Random RTP Port in all content FIRMWARE VERSION 1 0 7 3 HT70
55. oes 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 Enables HT70X 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 HT70X 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 If 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 to 15 or 20 minutes The URL for the HTTP HTTPS server used for firmware upgrade and configuration via 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 HT70X will only do HTTP HTTPS download once at boot up IP address or domain name of firmware server IP address or domain name of configuration server The password used for encrypting the XML configuration file using Ope
56. on defines different implementation of support 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 Define the Start Tone and Stop Tone 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 HT70X 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 HT70X supports a Duration Range from 100 to 10000 ms Default is Yes If set to No FLASH button could only be used for terminating calls Time period when the cradle is pressed Hook Flash to simulate FLASH To prevent FIRMWARE VERSION 1 0 7 3 HT70X USER MANUAL Page 43 of 52 On Hook Timing Gain Disable Line Echo Canceller LEC Disable Network Echo Suppressor Outgoing Call Duration Limit Ring Frequency Enable High Ring Power Ring Tones SIP Use ID Authenticate ID Password Andstream Innovative IP Voice amp Video unwanted activation of the Flash Hold and automatic phone ring back adjust this time value Default values are 300 minimum an
57. onfiguration HT701 and HT702 FXS SIP account s have its own configuration page HT704 has two 2 profiles for the four 4 FXS ports Admin Password Layer 2 QoS STUN Server is Keep alive interval TABLE 10 ADVANCED SETTINGS This contains the password to access the Advanced Web Configuration page This field is case sensitive Only the 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 802 1Q VLAN Tag 0 4094 SIP 802 1p 0 7 RTP 802 1p 0 7 IP address or Domain name of the STUN server This parameter specifies how often the HT70X 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 FIRMWARE VERSION 1 0 7 3 HT70X USER MANUAL Page 30 of 52 Use STUN to detect network connectivity Use DNS to detect network connectivity Firmware Upgrade and Provisioning Via TFTP Server Via HTTP HTTPS Server Firmware Server Path Config Server Path XML Config File Password HTTP HTTPS User Name HTTP HTTPS Password Andstream Innovative IP Voice amp Video 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 d
58. ovative IP Voice amp Video Vocoder is entered by choosing the appropriate option in Choice 1 The last Vocoder is entered by choosing the appropriate option in Choice 8 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 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 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 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 min 20ms max 200ms initial 50 5 min 10ms max 100ms This opti
59. rval before expiration which is half of the Register Expiration time or 10min maximum Retry registration if the process failed Default is 20 seconds Input value for SIP DSCP Diff Serv value in decimal default 24 And value for RTP DSCP Diff Serv value in decimal default 46 Defines the local SIP port the HT70X will listen and transmit The default value for FXS port is 5060 Defines the local RTP RTCP port pair the HT70X will listen and transmit It is the base RTP port for channel 0 When configured channel 0 uses this port value for RTP and the port 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 HT70X 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 HT70X are behind the same NAT 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 this will create a Semi Attendant Transfer When set to Yes device c
60. s of HT70X user manuals Only major new features or major document updates are listed here Minor updates for corrections or editing are not documented here CHANGES FROM 1 0 6 1 USER MANUAL e Added the options to enable disable NEC Disable Network Echo Suppressor e Added the options to restrict outgoing call duration Outgoing Call Duration Limit e Added the options to configure Ring Frequency e Added the options to Enable High Ring Power to support device consumer more power from FXS port e Added the options to enable disable Generate Continuous RFC2833 Events e Added the options to configure SIP REGISTER Contact Header Uses its WAN or LAN address e Added the options to Disable Multiple m line in SDP e Added the options to configure SIP Timer D e Added the options to configure Country Specific Deployment CHANGES FROM 1 0 5 2 USER MANUAL e Added the options to enable disable Always send HTTP Basic Authentication Information e Added the options to enable disable Do Not Escape as 23 in SIP URI e Rearrange Layer 2 QoS setting options Layer 2 QoS and move Layer 3 QoS to FXS Port page CHANGES FROM 1 0 4 14 USER MANUAL e Added the option to define Voice Frames per TX e Added the options to enable disable Use P Preferred Identity Header and Use Privacy Header Added the option to enable disable Add Auth Header On Initial REGISTER CHANGES FROM 1 0 4 8 USER MANUAL e Added the option t
61. ses or 3 Both HT70X and other VoIP Device can be connected through a router using public or private IP addresses with necessary port forwarding or DMZ HT70X 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 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 7 3 HT70X USER MANUAL Page 19 of 52 andstream Innovative IP Voice amp Video b Ifthe 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 itis configured as a send key or wait for 4 seconds NOTE When completing direct IP call the Use Random SIP RTP Port should set
62. sword for the ACS to connect to this device Password CPE SSL Certificate The Cert File for the phone to connect to the ACS via SSL CPE SSL Private Key Cert Key for the phone to connect to the ACS SSL Country Specific Default is None If choose any other option FXS related settings will be applied to most commonly used value in the configured the region Deployment System Ring Cadence Configuration option is set ring cadence on all FXS ports for all incoming calls Syntax c on1 off1 on2 off2 on3 off3 only cadences maximum Default is set to c 2000 4000 US standards Call Progress Tones 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 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 Note freq 0 4000Hz vol 30 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 Dir
63. 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 7 3 HT70X USER MANUAL Page 18 of 52 Feandstream Innovative IP Voice amp Video Examples 1 Dial 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 prefix 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 HT70X and other VoIP Device have public IP addresses or 2 Both HT70X and other VoIP Device are on the same LAN using private IP addres
64. 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 system up time since the last reboot Indicates whether the PPPoE connection is up if the HT70X is connected to DSL modem This filed indicates the type of NAT connection used by the HT70X FIRMWARE VERSION 1 0 7 3 HT70X USER MANUAL Page 29 of 52 Port Status andstream Innovative IP Voice amp Video 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 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 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 c
65. urn 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 HT70X as it may cause damage to the products and void the manufacturer warranty FIRMWARE VERSION 1 0 7 3 HT70X USER MANUAL Page 7 of 52 andstream Innovative IP Voice amp Video e This document contains links to HT70X GUI Interfaces Please download these examples for your reference here http www grandstream com products ht_series ht701 documents ht70x_qui zip e This document is subject to change without notice The latest electronic version of this user manual is available for download at http www grandstream com products ht series ht701 documents ht70x usermanual english pdf 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 7 3 HT70X USER MANUAL Page 8 of 52 andstream Innovative IP Voice amp Video CONNECT YOUR HT70X Connecting the HT70X is easy Before you begin please verify the contents of the HT70X package EQUIPME
66. y 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 FIRMWARE VERSION 1 0 7 3 HT70X USER MANUAL Page 13 of 52 DHCP Server Client Telnet Server Fax over Ip QoS Transport Protocol DTMF Method IP Signaling Provisioning Control Device Management Dial Plan Caller ID Call Handling Features andstream Innovative IP Voice amp Video packet header 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 tra
67. ype of firewall NAT If the detected NAT is a Full Cone Restricted Cone or a Port Restricted Cone the HT70X 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 HT70X will periodically every 20 seconds or so send a blank UDP packet with no payload data to the SIP server to keep the hole on the NAT open SIP User ID User account information provided by VoIP service provider ITSP Usually in the form of digit similar to phone number or actually a phone number HT701 and HT702 only Authenticate ID SIP service subscribers Authenticate ID used for authentication Can be identical to or different from SIP User ID HT701 and HT702 only Authenticate Password service subscriber s account password HT701 and HT702 only Name SIP service subscriber s name for Caller ID display HT701 and HT702 only DNS Mode 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 Tel URI The default setting is Disabled If the phone has an assigned PSTN Number this field should be set to UserzPhone then a User Phone

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