Grandstream Networks, Inc.
Contents
1. 13 SUPPORT 13 GALL FEATURES esis es eds ees ccc 14 CONFIGURATION GUIDE scicacsisi cst 15 CONFIGURING THE 701 THROUGH VOICE 0020400000 000 0 15 CONFIGURING THE 701 VIA WEB BROWSER 15 IMPORTANT 0000400 16 Preferred VOCODER Codec 16 ADVANCED USER CONFIGURATION 19 SAVING THE CONFIGURATION CHANGES 28 REBOOTING THE HT 701 FROM 28 CONFIGURATION THROUGH A CENTRAL 28 42225 tac 29 FIRMWARE UPGRADE THROUGH 805 29 CONFIGURATION FILE DOWNLOAD 30 FIRMWARE AND CONFIGURATION FILE PREFIX AND POSTFIX 30 MANAGING FIRMWARE AND CONFIGURATION FILE DOWNLOAD 30 RESTORE FACTORY DEFAULT SETTING 02 31 Grandstream Networks Inc HT 701 User Manual Pa2. A wants to Blind Transfer B to 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 confirm tone Then A can hang up NOTE Enable Call Feature must be set to Yes web configuration page Caller A can place a call on hold and wait for one of three situations 1 A quick confirmation tone similar to call waiting tone followed by a dial tone This indicates the transfer is successful transferee has received a 200 OK from transfer target At this point Caller A can either hang up or make another call 2 A quick busy tone followed by a restored call on supported platforms only This means the transferee has received a 4xx response for the INVITE and we will try to recover the call The busy tone is just to indicate to the transferor that the transfer has failed 3 Continuous busy tone The phone has timed out Note continuous busy tone does not indicate the transfer has been successful nor does it indicate the transfer has failed It often means there was a failure to receive second NOTIFY check firmware for most recent release Grandstream Networks Inc HT 701 User Manual Page 12 of 31 Firmware Version 1 0 0 17 Last Updated 02 2012 ndstream Innovative IP Voice amp Video Attended Transfer Assume that Caller A and B are in conversation Caller A wants to Attend Transfer
3. 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 Grandstream Networks Inc HT 701 User Manual Page 13 of 31 Firmware Version 1 0 0 17 Last Updated 02 2012 ndstream Innovative IP Voice amp Video CALL FEATURES The HT701 supports all the traditional and advanced telephony features Table 7 HT701 Call Features CallFeatures 02 Forcing a Codec per call 027110 PCMU 027111 PCMA 02723 G723 02729 G729 0272616 G726 r16 0272624 G724 r24 0272632 G726 r32 0272640 G726 r40 027201 iLBC 03 Disable LEC call Dial 03 number dial tone is played in the middle 16 Enable SRTP 17 Disable SRTP 30 Block Caller ID for all subsequent calls 31 Send Caller ID for all subsequent calls 47 Direct IP Calling Dial 47 IP address No dial tone is played 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
4. Preferred Vocoder ndstream Innovative IP Voice amp Video e 011 2 9 x allows international calls starting with 011 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 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 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 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
5. 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 HT701 and other VoIP Device have public IP addresses or 2 Both HT701 and other VoIP Device are on the same LAN using private IP addresses or 3 Both HT701 and other VoIP Device can be connected through a router using public or private IP addresses with necessary port forwarding or DMZ HT701 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 Grandstream Networks Inc HT 701 User Manual Page 11 of 31 Firmware Version 1 0 0 17 Last Updated 02 2012 ndstream Innovative IP Voice amp Video 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 Exam
6. AOL s Netscape or Mozilla Firefox installed on Windows or Unix OS Macintosh OS is included Access the Web Configuration Menu 1 Find the IP address of the HT701 using voice prompt menu option 02 2 Open a web browser type the IP address You will see the log 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 Grandstream Networks Inc HT 701 User Manual Page 15 of 31 Firmware Version 1 0 0 17 Last Updated 02 2012 ndstream Innovative IP Voice amp Video 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 PORTs configuration pages Please reference the GUI pages using the following link http Awww grandstream com products ht_series HT701 documents HT701_ qgui zip NOTE If you cannot log into the configuration page by using the default password p
7. 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 If Caller C does not answer the call Caller can press flash to resume call with Caller B NOTE When Attended Transfer fails and A hangs up the HT701 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 supports Bellcore style 3 way Conference Instructions for 3 way conference Assume that call party A and B are in conversation Caller A HT701 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 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 If A hangs up the conference will be terminated for all three parties when configuration Transfer on Conference Hangup is set to No If the configuration is set to Yes A will transfer to so that B and C can continue the conversation oar Fax Support HT701 supports FAX in two modes 1 T 38
8. 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 0 17 e g 72 172 83 110 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 www grandstream com support firmware Currently the HTTP firmware server IP address is 72 172 83 110 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 HT701 will attempt to retrieve the new image files by downloading them into the GXW400x s SRAM During this stage the HT701 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 HT701 will stop the TFTP HTTP HTTPS process and simply boot using the existing code ima
9. IP Voice 8 Video Syslog Level Select the HT701 to report the log level Default is NONE The level is one of DEBUG INFO WARNING or ERROR Syslog messages are sent based on the following events 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 0 memory exception ERROR level 9 GS Gl as Eo 0 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 Download Device Allows user to download and save a text file containing all the P values of each setting as Configuration configured at that point the unit Note For Security Reasons all Passwords won t be Downloaded Upload Firmware Allows the user to upgrade the firmware with the single firmware file by browsing it and loading it from your computer local directory Table 11 Account Settings Profile Active Primary SIP Server Failover SIP Server Prefer Primary SIP Server Outbound Proxy SIP transport NAT Traversal STUN SIP User ID When set to
10. 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 INVITE The HT701 supports up to 5 different Vocoder types including G 711 A U law G 726 Supports bit rates 16 24 32 and 40 G 723 1 G 729A B E iLBC and AAL2 The user can configure Vocoders in a preference list that will be included with the same preference order in SDP message The first 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 Grandstream Networks Inc HT 701 User Manual Firmware Version 1 0 0 17 Page 26 of 31 Last Updated 02 2012 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 Polarity Reversal Loop Current Disconnect Loop Current Disconnect Duration Hook Flash Timing On Hook Timing Gain Disable Line Echo Canceller LEC Ring Tones ndstream Innovative IP Voice amp Video Default is 6 3kbps Defines the encoding rate for G 723 vocoder Sets the iLBC frame size in 20ms or 30ms Defines payload type for iLBC Default value is 97 The valid
11. amp Video Via TFTP Server This is the IP address of the configured TFTP server If selected and it is non zero or not blank the HT701 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 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 HT701 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
12. of Voice Mails Press 9 to reboot the device Enter MAC address to restore factory default setting See Restore Factory Default Setting section Automatically returns to main menu Grandstream Networks Inc HT 701 User Manual Firmware Version 1 0 0 17 Page 10 of 31 Last Updated 02 2012 ndstream Innovative IP Voice amp Video 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 tk 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 an option 4 All entered digit sequences have known lengths 2 digits for menu option and 12 digits for IP address For 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 can not be deleted but the phone may prompt error once it is detected Placing a Phone Call Phone or Extension Numbers 1 Dial the number directly and wait for 4 seconds Default No Key Entry Timeout 2 Dial the number directly and press Use as dial key must be configured in web configuration 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
13. of voice codecs Under Profile web pages choose your preferred order of different codecs e or G711p a Grandstream Networks Inc HT 701 User Manual Page 16 of 31 Firmware Version 1 0 0 17 Last Updated 02 2012 Innovative IP Voice 8 Video e 729 A B E G723 e 2726 16 24 32 40 e iLBC AAL2 all G 726 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 IP Address There are two modes to operate the HT701 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 HT701 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 DNS Server 1 primary DNS Server 2 secondary fields These fields are set to zero by default DHCP hostname Default is blank This option specifies the name of the client This field is optional but may be required by some Internet Service Providers DHCP domain Default is blank This option specifies the domain name that client
14. should use when resolving hostnames via the Domain Name System DHCP vendor class ID Default is HT7XX Used by clients and servers to exchange vendor specific information PPPoE account ID PPPoE username Necessary if ISP requires you to use a PPPoE Point to Point Protocol over Ethernet connection PPPoE password PPPoE account password PPPoE Service Name Default is blank This field is optional If your ISP uses a service name for the PPPoE connection enter the service name here Preferred DNS server The preferred DNS Server to be used Time Zone Controls how the date time is displayed according to the specified time zone Grandstream Networks Inc HT 701 User Manual Page 17 of 31 Firmware Version 1 0 0 17 Last Updated 02 2012 ndstream Innovative IP Voice amp Video Self Defined Time Zone The syntaxis 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 iteration of the weekday 1st Sunday 3rd Tuesday etc 3 number indicates Weekday 0 1 2 6 for Sun Mon Tue Sat Therefore this example is
15. t come from the SIP Messages from SIP proxy they will be rejected If this option is enabled the device will not be able to make Proxy Only direct IP calls SIP T1 Timeout 1 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 SIP T2 Interval Maximum retransmission interval for non INVITE requests and INVITE responses DTMF Payload Type Sets the payload type for using RFC2833 Preferred method The 701 supports up to different methods including in audio via RTP RFC2833 and via Sip Info The user can configure method a priority list Disable DTMF Default is No If set to yes use above DTMF order without negotiation Negotiation DTMF via RFC2833 Send DTMF via RTP According to RFC 2833 DTMF via SIP INFO Send DTMF via SIP INFO message Send Flash Event Default is No If set to yes flash will be sent as DTMF event Enable Call Features Default is Yes If Yes call features using star codes will be supported locally Offhook Auto Dial 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 HT701 will automatically append the and the host portion of the corresponding SIP address Proxy Require SIP Extension to notify SIP server that the unit is behind
16. the DST which starts from the second Sunday of March to the 1st Sunday of November Allow DHCP server to Default No 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 that located on FXS port 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 push 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 9 Status Page MAC Address The device ID in HEX format This is very important ID for ISP troubleshooting LAN Mac address will appear in this place The LAN 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 LAN IP Address This field shows the LAN IP address of the HT701 Product Model This field contains the product model info Hardware Version This field shows the hardware revision of the unit and the part number Software Version Program This is the main software release This number is always used for firmware upgrade Current release is 1 0 0 17 Boot and Loader are seldom changed Bootloader curren
17. 000 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 4000 2 vol 30 0dBm 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 NTP server URI or IP address of the NTP Network Time Protocol server This parameter synchronizes the date and time Allow DHCP option 42 Default NO 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 Grandstream Networks Inc HT 701 User Manual Page 21 of 31 Firmware Version 1 0 0 17 Last Updated 02 2012 Innovative
18. 6000 1 1 G Yes daily at hour uT G Yes weekly on day 771 Grandstream Networks Inc HT 701 User Manual Page 30 of 31 Firmware Version 1 0 0 17 Last Updated 02 2012 ndstream Innovative IP Voice amp Video RESTORE FACTORY DEFAULT SETTING WARNING 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 two 2 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 Insert pin 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 Dial for voice prompt Enter 99 and wait for reset voice prompt Enter the encoded MAC address Look below on how to encode MAC address Wait 15 seconds and device will automatically reboot and restore factory settings PON 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 inthe MAC
19. 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 followed 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 then hang up Flash Hook Toggles betwe
20. Ae Innovative IP Voice amp Video Grandstream Networks Inc HT701 Analog Telephone Adaptor HT701 User Manual www grandstream com Firmware Version 1 0 0 17 http esupport grandstream com TABLE OF CONTENTS HT701 User Manual ndstream Innovative IP Voice amp Video pT et Seg a a ee eae ee ee 4 SAFETY 6 24 4 NV ARRAN TY 4 a ce a 5 2 5 01595 00 chan 5 5 8 SOFTWARE FEATURES OVERVIEW 2 4 444 44 8 HARDWARE 9 BASIC OPERATIONS 2222 eed eae 10 UNDERSTANDING 701 VOICE PROMPT 10 PLACING PHONE 11 12 CALE WAITING 12 THRAN SPER 12 3 CONFERENCING
21. ED 04 Voice mail waiting for Line X Phone 1sec OFF LED 05 Device has normal WAN connection and has obtained IP internet ON address LED 06 Internet link error Device is powered up and ready to connect to the Internet but the WAN INTERNET port is Internet OFF down LED 07 Internet DHCP Error Device is properly connected but it is unable to retrieve an IP address from the device it is Internet 0 25sec ON 0 25sec OFF connected to LED 08 Line Registration failed Device is properly setup can connect to provider s network but cannot register to Phone OFF provider s SIP proxy no 200 OK LED 09 Device is connected has physical data link but there are incorrect network settings typically associated with PPPoE Internet 0 25sec ON 0 25sec OFF connection failure LED 10 Hazardous potential test failed Hazardous AC or DC voltage is present on the tip and ring or both signals of Phone 0 25sec ON 0 25sec OFF phone line X LED 11 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 ON voltage on the FXS phone line LED 12 Resistive fault test failed Either tip or ring is shorted to Phone 0 25sec 0 25sec OFF ground or they are shorted to each other LED 13 Receiver off hook test fail One or more phones are off Phone 0 25sec 0 25sec OFF hook on phone line during test LED 14 REN test
22. HTTPS server Name HTTP HTTPS The password needed to authenticate with the HTTP HTTPS server Password Firmware File Prefix 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 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 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 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 HT701 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 Key Used for firmware encryption Sh
23. ION GUI INTERFACE EXAMPLES HT701 USER MANUAL http www grandstream com products ht_series ht701 documents ht701_qui zip 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 Grandstream Networks Inc HT 701 User Manual Page 3 of 31 Firmware Version 1 0 0 17 Last Updated 02 2012 ndstream Innovative IP Voice amp Video WELCOME Thank you for purchasing Grandstream s HT701 the affordable feature rich Analog Telephone Adaptor Grandstream HandyTone 701 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 of the HT701 including one FXS port with an independent SIP account This manual will help you learn how to operate and manage your HandyTone701 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 new provisioning support among other features This HT701 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 HT701 phone complies with FCC CE and various safety standards The HT701 power adaptor is
24. IP account amp profile Supports Voice Codecs G 711 G 723 G 726 32 G 729 T 38 Fax Comprehensive Dial Plan support for Outgoing calls G 168 Echo Cancellation Voice Activation Detection Concealment PLC Supports PSTN PBX analog telephone sets or analog trunks VAD Comfort Noise Generation CNG and Packet Loss Table 4 HT701 Technical Specifications Telephone Interfaces Network Interface LED Indicators Reset Button Voice over Packet Capabilities Voice Compression Telnet Server Fax over IP QoS IP Transport DTMF Method IP Signaling Provisioning Control 1 FXS port 1 SIP account 1 RJ45 for LAN 10 100 Base TX Full Duplex Power INTERNET LINK ACTIVITY PHONE Factory Reset button Voice Activity Detection VAD with CNG comfort noise generation and PLC packet loss 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 G 711 Annex PLC Annex VAD CNG format encoder and decoder G 723 G 726 ADPCM G 729 iLBC G 726 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 Yes T 38 compliant Grou
25. 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 either to continue the conversation or hang up Default is No this will create a Semi Attendant Transfer When set to Yes device can transfer the call upon receiving ring back tone or SIP message 180 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 Grandstream Networks Inc HT 701 User Manual Page 23 of 31 Firmware Version 1 0 0 17 Last Updated 02 2012 ndstream Innovative IP Voice amp Video Support SIP Instance ID 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 Validate incoming SIP Default is No If set to yes all incoming SIP messages will be strictly validated message according to RFC rules If message will not pass validation process call will be rejected Check SIP User ID for Default is No Check the incoming SIP User ID in Request URI If they don t match the incoming INVITE call will be rejected If this option is enabled the device will not be able to make direct IP calls Allow Incoming SIP Default is No Check the incoming SIP messages If they don
26. Server IP Address Upgrade Protocol Firmware Version Firmware Upgrade Direct IP Calling Voice Mail RESET Invalid Entry Press for the next menu option Press 4 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 Mode configure the IP address information using menus 02 to 05 If using Dynamic IP Mode all address information comes from the DHCP server automatically after reboot The current WAN IP address is announced If using Static Mode enter 12 digit new IP address You need to reset the HT to take affect the new IP address Same as menu 02 Same as menu 02 Same as menu 02 Press 9 to move to the next selection in the list e PCMU PCMA e iLBC G 726 e G 723 G 729 Announces the Mac address of the unit Announces current Firmware Server IP address Enter 12 digit new IP address Announces current Config Server Path IP address new IP address Enter 12 digit Upgrade protocol for firmware and configuration update Press 9 to toggle between TFTP HTTPS Firmware version information Firmware upgrade mode Press 9 to toggle among the following three options always check check when pre suffix changes never upgrade Enter the target IP address to make a direct IP call after dial tone See Make a Direct IP Call Number
27. Yes the FXS port is activated SIP Server s IP address or Domain name provided by VoIP service provider Failover SIP Servers 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 HT701 for firewall or NAT penetration in different network environments lf symmetric NAT is detected STUN will not work and ONLY outbound proxy can correct the problem User can select UDP or TCP or TLS This parameter defines whether not the 701 NAT traversal mechanism is activated If activated by choosing Yes and a STUN server is also specified then the HT701 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 HT701 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 HT701 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 User account information provided by VoIP service provider ITSP Usually in the form of digit similar to phone number or act
28. address Use the following mapping 0 9 0 9 A 22 press the 2 key twice will show on the LCD B 222 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 000282006395 it should be keyed as 0002228200333395 NOTE 1 Factory Reset will be disabled if the Lock keypad update is set to Yes 2 Please be aware by default the HT701 WAN side HTTP access is disabled After a factory reset the device s web configuration page can be accessed only from its LAN port 3 If the HT701 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 Grandstream Networks Inc HT 701 User Manual Page 31 of 31 Firmware Version 1 0 0 17 Last Updated 02 2012
29. age Table 10 Advanced Settings Admin Password Layer 3 QoS Layer 2 QoS STUN Server Keep alive interval Use STUN to detect network connectivity Firmware Upgrade and Provisioning 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 This field defines the layer 3 QoS parameter which can be the value used for IP Precedence or Diff Serv or MPLS Default value is 48 Value used for layer 2 VLAN tag Default setting is blank IP address or Domain name of the STUN server This parameter specifies how often the HT701 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 Enables HT701 to download firmware or configuration file through either the TFTP HTTP or HTTPS server Grandstream Networks Inc HT 701 User Manual Page 19 of 31 Firmware Version 1 0 0 17 Last Updated 02 2012 ndstream Innovative IP Voice
30. ch voltage drop described in topic above 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 On hook timing is the minimum time for an on hook event to be validated Voice path volume adjustment e xis again level for signals transmitted by FXS e Tx is again level for signals received 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 This function lets you configure ring tone cadence preferences User has 10 choices Grandstream Networks Inc HT 701 User Manual Page 27 of 31 Firmware Version 1 0 0 17 Last Updated 02 2012 ndstream Innovative IP Voice amp Video The configuration completed in Distinctive Ring Tones block in the same page applies to ring tones cadences configured here Sav
31. compliant with UL standard Only use the universal power adapter provided with the HT701 package The manufacturer s warranty does not cover damages to the phone caused by unsupported power adaptors Warranty If you purchased your HT701 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 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 HT701 as it may cause damage to the products and void the manufacturer warranty e This document contains links to HT701 GUI Interfaces Please download these examples from http www grandstream com products ht_series ht701 documents ht701 qui zip for your reference e This document is subject to change without notice The latest electronic version of this user manual is available for download at http Awww grandstream com products ht_series ht701 documents ht701_ usermanual_english pdf Reproduction or transmittal of the entire or any part in any form or by any mean
32. en 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 Grandstream Networks Inc HT 701 User Manual Page 14 of 31 Firmware Version 1 0 0 17 Last Updated 02 2012 ndstream Innovative IP Voice amp Video CONFIGURATION GUIDE Configuring the HT701 through Voice Prompts DHCP Select voice menu option 01 to enable HT701 to use DHCP STATIC Select voice menu option 01 to enable HT701 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 HT701 Via Web Browser HT701 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 HT701 through a web browser such as Microsoft s IE
33. f server supports TEL URI format then this option needs to be selected Controls whether the HT701 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 HT701 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 HT701 will listen and transmit The default value for FXS port is 5060 Defines the local RTP RTCP port pair the HT701 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_value 1 for its RTCP The default value for FXS port is 5004 Default is No This parameter forces the random generation of both the local SIP and RTP ports when set to Yes This is usually necessary when multiple HT701 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
34. failed high REN detected Too many parallel Phone 0 25sec 0 25sec OFF phones connected to phone line X LED 15 Line is active Phone 1 sec ON 1 sec OFF Grandstream Networks Inc HT 701 User Manual Page 6 of 31 Firmware Version 1 0 0 17 Last Updated 02 2012 ndstream Innovative IP Voice amp Video LED 16 Line inactive Phone ON LED 17 During Provisioning Stage Internet 0 2sec ON 0 2sec OFF Phone LED 18 During Firmware Recovery Stage 0 4sec ON 0 4sec OFF Note In Provisioning and Firmware Recovery Stage the power LED is Steady ON Figure 2 HT701 Connection Diagram Cordless Phone gt LAN FXS Internet Analog ADSL Cable Modem Ethernet Fax Grandstream Networks Inc HT 701 User Manual Page 7 of 31 Firmware Version 1 0 0 17 Last Updated 02 2012 ndstream Innovative IP Voice amp Video PRODUCT OVERVIEW The HT701 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 HT701 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 G 723 G 726 32 G 729 and iLBC Software Features Overview 1 S
35. ge 2 of 31 Firmware Version 1 0 0 17 Last Updated 02 2012 ndstream Innovative IP Voice amp Video TABLE OF FIGURES HT701 USER MANUAL FIGURE 1 CONNECTING THE H T70 Tarrio 5 FIGURE 2 HT 701 CONNECTION E EREE AE E RE Eia 7 TABLE OF TABLES HT701 USER MANUAL TABLE 1 DEFINITIONS OF THE HT701 CONNECTORS 5 TABLE 2 DEFINITIONS OF THE HT701 LEDS woo cccccecesccecceecesccesseecessceeseecesscesssecesseecseeceasecensecssseeeseeseseeenaees 6 TABLE 3 701 TECHNICAL SPECIFICATIONS 0 cccccccecessceessecessceeseecesscecssecessceeseeceascecssecesssecssecessesenseceaseeesaeensesenaees 8 TABLE 5 HT701 IVR MENU DEFINITIONS eee eee ceeececeseceecceceseceeceecuecesseecusceeseecascesseecascesseecnascesseecsaseeeseeenseeesaes 10 TABLE 6 AT 701 CALL FEATURES 14 TABLE 72 17 TABLE 8 STATUS 18 TABLE 9 ADVANCED 19 STABLE 1 02 ACCOUNT SETTINGS cdsdvatecdiveevacvadeccdesiySetelass 22 CONFIGURAT
36. ge 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 LAN It is recommended to conduct firmware upgrade 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 e Grandstream s latest firmware is available at http Awww grandstream com support firmware Oversea users are strongly recommended to download the binary files and upgrade firmware locally in a controlled LAN environment e Alternatively user can download a free TFTP or HTTP server and conduct local firmware upgrade 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 y firmware htm Instructions for local firmware upgrade 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 Grandstream Networks Inc HT 701 User Manual Page 29 of 31 Firmware Version 1 0 0 17 Last Updated 02 2012 ndstream Innovative IP Voice amp Video 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 Star
37. ge waiting Do Not Disturo DND call return service Bellcore Type 1 amp 2 ETSI BT NTT and DTMF based CID Yes EN55022 EN55024 and FCC part15 Class B UL Hardware Specification The table below lists the hardware specification of HT701 TABLE 5 HT701 HARDWARE SPECIFICATION LAN Interface LED Universal Switching Power Adaptor 1 RJ45 10 100Mbps 4 LEDs GREEN Input 100 240V AC 50 60Hz 0 18A Max Output 12V DC 0 5A UL certified Dimension 86mm L x 65mm W x 25mm H Weight 31 g 0 07165 32 104 F 0 40 Humidity 10 90 non condensing Compliance FE Grandstream Networks Inc HT 701 User Manual Page 9 of 31 Firmware Version 1 0 0 17 Last Updated 02 2012 BASIC OPERATIONS Understanding HT701 Voice Prompt Innovative IP Voice 8 Video HT701 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 HT701 IVR Menu Definitions Voice Prompt Main Menu 01 02 03 04 05 07 10 13 14 15 16 17 47 86 99 Enter a Menu Option DHCP Mode Static IP Mode IP Address IP address Subnet IP address Gateway address DNS Server IP address Preferred Vocoder MAC Address Firmware Server IP Address Configuration
38. h 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 e Example 3 1 2 9 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 1900 lt 1617 gt 2 9 xxxxxx 1 2 9 2 9 011 2 9 x 3469 11 Explanation of example rule reading from left to right e 41900x prevents dialing any number started with 1900 e 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 Grandstream Networks Inc HT 701 User Manual Page 25 of 31 Firmware Version 1 0 0 17 Last Updated 02 2012 Subscribe for MWI Send Anonymous 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 Use First Matching Vocoder in 2000K SDP
39. in seconds The default value is 90 seconds If selecting Yes the phone will use session timer when it makes outbound calls if remote party supports session timer If selecting Yes the phone will use session timer when it receives inbound calls with session timer request 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 As a Caller select UAC to use the phone as the refresher or UAS to use the Callee or proxy server as the refresher 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 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 If set to Yes Enables the use of PRACK Provisional Acknowledgment method Default is No
40. ing the Configuration Changes Click the Update button in the Configuration page to save the changes to the HT701 configuration The following screen confirms that the changes are saved Reboot or power cycle the HT701 to make the changes take effect Rebooting the HT701 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 HT701 can be automatically configured from a central provisioning system When HT701 boots up it will send TFTP or HTTP HTTPS request to download configuration file cfg000b82xxxxxx 000082 where 000b82xxxxxx is the LAN MAC address of the HT701 It will first request cfg000b82xxxxxx then cfg000b82xxxxxx xml The configuration file can be downloaded via TFTP or HTTP HTTPS from the central server A service provider or an enterprise with large deployment of HT701 can easily manage the configuration and service provisioning of individual devices remotely from a central server Grandstream has a provisioning system called GAPS Grandstream Automated Provisioning System that is used to support automated configuration of Grandstream devices GAPS uses enhanced NAT friendly TFTP or HTTP thus no NAT issues and other communication protocols t
41. lease check with 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 gateway 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 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 Profile web pages Set this to Yes when gateway is behind firewall on a private network DTMF Methods DTMF Settings are in FXS portX pages DTMF in audio e DTMF via RTP RFC2833 e DTMF via SIP INFO Set priority of methods according to your preference This setting should be based on your server DTMF setting PREFERRED VOCODER Copec The HT701 supports a broad range
42. not be provided to analog phone Caller ID connected to this FXS port Disable Call Waiting Default is No This is to disable the stutter Call Waiting Tone when a Call Waiting Tone information arrives The CWCID information will still be displayed Disable Receiver Default is No If set to yes it will disable the warning to alert that the phone has been Offhook Tone left off hook for an extended period of time Grandstream Networks Inc HT 701 User Manual Page 24 of 31 Firmware Version 1 0 0 17 Last Updated 02 2012 Disable Reminder Ring for On Hold Call Disable Visual MWI Ring Timeout Delayed Call Forward Wait Time No Key Entry Timeout Early Dial Dial Plan Prefix Use as Dial Key Dial Plan Innovative IP Voice amp Video Default is No Do not play the reminder ring when this is set to Yes If set to Yes the MWI information will not be transferred to the analog phone connected to the FXS port Incoming call will stop ringing when not picked up given a specific period of time Default value is 20 seconds 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
43. o communicate with each individual Grandstream device for firmware upgrade remote reboot etc Grandstream provides GAPS service to VolP 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 customers TFTP HTTP HTTPS server for further provisioning Grandstream also provide GAPSLITE software package which contains our NAT friendly TFTP server and a configuration tool to facilitate the task of generating device configuration files The GAPSLITE configuration tool is now free to end users The tool and configuration template are available for download from http www grandstream com support tools Grandstream Networks Inc HT 701 User Manual Page 28 of 31 Firmware Version 1 0 0 17 Last Updated 02 2012 ndstream 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
44. ould be 32 digit in hexadecimal representation End user should keep it blank Authenticate Conf File If set to Yes config is authenticated before acceptance This protects the configuration from an unauthorized change Grandstream Networks Inc HT 701 User Manual Page 20 of 31 Firmware Version 1 0 0 17 Last Updated 02 2012 Innovative IP Voice 8 Video Firmware Key Used for firmware encryption Should be 32 digit in hexadecimal representation End user should keep it blank SSL Certificate The user specify SSL certificate used for SIP over TLS in X 509 format SSL Private Key The user specify SSL private key used for SIP over TLS in X 509 format SSL Private Key User specify password to protect the private key above Password ACS URL User specify the Auto Configuration Server s URL TR 069 protocol ACS Username User specify the ACS Username ACS Password User specify the ACS password Periodic Inform Default is No If set to YES device will send inform packets to the ACS Enable Inform Frequency that the inform packets will be sent out to the ACS nterva Connection Request Set a user name for the ACS to connect to this device Username Connection Request a password for the ACS to connect to this device Password System Ring Cadence 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 4
45. p 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 TLS SIPS SIP over TCP TLS Grandstream Networks Inc HT 701 User Manual Firmware Version 1 0 0 17 Page 8 of 31 Last Updated 02 2012 Device Management Dial Plan Universal Switching Power Adaptor Environmental Dimensions H x W x D Short Haul Loop Call Handling Features Caller ID Polarity Reversal Wink EMC Safety ndstream Innovative IP Voice amp Video 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 Yes Input 100 240 VAC 50 60 Hz 0 18A Output 12VDC 0 5A UL certified Operational 32 104 F or 0 40 C Storage 10 130 F Humidity 10 90 Non condensing 86mm L x 65mm W x 26mm H 5REN Up to 1Km on 24 AWG wire Caller ID display or block Call waiting caller ID Call waiting flash Call transfer hold forward 3 way conferencing messa
46. ples 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 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 it is configured as a send key or wait for 4 seconds NOTE When completing direct IP call the Use Random 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 in conversation
47. 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 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 can 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 at least 2 digits number xx at least 2 digit number exclude 3 5 any digit of 3 4 or 5 147 any digit 1 4 or 7 lt 2 011 gt replace digit 2 with 011 when dialing lt 1 gt add a leading 1 to all numbers dialed vice versa will remove from the number dialed or a e o o o o o o o Example 1 369 11 1617xxxxxxx Allow 311 611 911 and any 11 digit numbers wit
48. ps Reset a Table 1 Definitions of the HT701 Connectors Power Cable Power adapter connection Internet Port RJ 45 Connect to the internal LAN network or router RESET Factory Reset button Press for 7 seconds to reset factory default settings FXS Port RJ 11 FXS port to be connected to analog phones fax machines Grandstream Networks Inc HT 701 User Manual Page 5 of 31 Firmware Version 1 0 0 17 Last Updated 02 2012 andstream Innovative IP Voice amp Video There are four 4 LED buttons that help you manage the status of your HandyTone Table 2 Basic Definitions of the HT701 LEDs Pattern SY i POWER LED Indicates Power Remains ON when power is connected Internet LED Indicates Access to Internet Remains ON while there is Access Link Activity LED Indicates if There is Activity on the Internet Port PHONE LED 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 display green when ON Table 3 Advanced Definitions of the HT701 LEDs Pattern Patten Condition LED Behavior Number LED 01 Device has normal power Power LED 02 Power Error Power is removed from the device or power Power OFF supply with improper voltage is plugged in LED 03 Line X is registered normally to the sip providers network and is ready to make a call Prone ON L
49. 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 Fax Pass Through if a fax tone is detected Select either Fixed or Adaptive based on network conditions Select Low Medium or High based on network conditions e High initial 200ms min 40ms max 600ms Note all vocoders can meet the high requirement e Medium initial 100 5 min 20ms max 200ms e initial 50ms 10ms max 100ms This option defines different implementation of support SRTP squired RTP transmission mode Dependent on standard phone type and location Bellcore Telcordia ETSI FSK ETSI DTMF SIN 227 BT amp NTT Japan 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 HT701 uses this method for signaling call termination Method initiates short voltage drop on the line when remote VoIP side disconnects an active call Here can be configured duration of su
50. s electronic or print for any purpose is not permitted without the express written permission of Grandstream Networks Inc Grandstream Networks Inc HT 701 User Manual Page 4 of 31 Firmware Version 1 0 0 17 Last Updated 02 2012 ndstream Innovative IP Voice amp Video CONNECT YOUR HT701 Equipment Packaging The HT701 ATA package contains HT701 Main Case e One Universal Power Adaptor e One Ethernet Cable Connecting the HT701 The HT701 is designed for easy configuration and easy installation Configure the HT701 following the directions in the Configuration section of this manual 1 Connect a standard touch tone analog telephone to the PHONE port 2 Insert a standard RJ11 telephone cable into the Phone1 port and connect the other end of the telephone cable to the analog telephone 3 Insert the Ethernet cable into the WAN port of HT701 and connect the other end of the Ethernet cable to an uplink port a router or a modem etc 4 Insert the power adapter into the HT701 and connect it to a wall outlet The HT701 Analog Telephone Adaptor is an all in one VoIP integrated device designed to be a total solution for networks providing VoIP services The HT701 VoIP features and functions are available using a regular analog telephone Figure 1 Connecting the HT701 HT 701 HT 701 Back View Front View hone lriterniet Port RJ 11 FXS Ports Display LEDs RJ 45 connector Power green 10 100 Mb
51. s 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 one directory In addition when the field Check New Firmware only when 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 Automatic Upgrade 10080 9 No G Yes every minutes 60 525
52. t 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 hitp 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 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 detailed parameter list please refer to the corresponding firmware release configuration template When Grandstream Device boots up or reboots it will issue request for configuration file named where iS the LAN MAC address of the device i e cfg000b820102ab The configuration file name should be in lower cases Firmware and Configuration File Prefix and Postfix Firmware Prefix and Postfix allow
53. t version is 1 0 0 xx Core current version 1 0 0 xx Base current version is 1 0 0 xx Software Status This field shows the status of the unit and its actual memory System Up Time Shows system up time since the last reboot PPPoE Link Up Indicates whether the PPPoE connection is up if the HT701 is connected to DSL modem Grandstream Networks Inc HT 701 User Manual Page 18 of 31 Firmware Version 1 0 0 17 Last Updated 02 2012 NAT Port Status Innovative IP Voice 8 Video This filed indicates the type of NAT connection used by the HT701 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 its SIP Server FXS Port user has set Do Not Disturb FXS Port user has set his calls to be forwarded unconditionally to ext 613 FXS Port user has not set Busy or Delay call Forward Advanced User Configuration 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 configuration 701 FXS SIP account has its own configuration p
54. the NAT Firewall Use NAT IP NAT IP address used in SIP SDP message Default is blank Distinctive Ring Tone 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 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 Disable Call Waiting Default is No If set to YES Call Waiting indication information will not be provided to analog phone connected to this FXS port Disable Call Waiting Default is No If set to YES Call Waiting caller ID will
55. ually a phone number Grandstream Networks Inc HT 701 User Manual Page 22 of 31 Firmware Version 1 0 0 17 Last Updated 02 2012 ndstream Innovative IP Voice amp Video Authenticate ID Authenticate Password Name DNS Mode Tel URI SIP Registration Unregister on Reboot Outgoing Call without Registration Register Expiration Registration Retry Wait Time Local SIP port Local RTP port Use Random Port Refer to Use Target Contact Transfer on Conference Hang up Enable Ring Transfer Disable Bellcore Style 3 Way Conference Remove OBP from Route Header 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 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 I
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