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Grandstream Networks 386 User's Manual

1. Waiting NAT Traversal STUN No Key Entry Timeout Preferred Vocoder es in listed order CHOICES choice 2 choice 3 choice 4 choice 5 choice 6 Voice Frames per TX G723 rate E to 10 20 32 64 for G711 G726 G723 other codecs respectively iLBC frame size G 20ms L 30ms iLBC payload type E cn 96 and 127 default is 97 Silence Suppression Fax Mode Early Dial a Yes use Yes only if proxy supports 484 response Dial Plan Prefix this prefix string is added to each dialed number Use as Dial Key Mi Yes if set to Yes will function as the Re Dial key SUBSCRIBE for MWI Send Anonymous Li Yes caller ID will be blocked 1f set to Yes Special Feature E Frequency1 Frequency2 ON x10ms OFF x10ms Distinct Ring Tones Hz Hz C1 C2 C3 C1 C2 C3 30 HandyTone 386 User Manual Grandstream Networks Inc Dial Tone Recall Dial Tone Message Waiting Audible Ringing Busy Tone Reorder Tone es H H H anata Receiver offhook Volume Amplification EE ae 7 The explanations provided apply to both of the FXS port configuration parameters SIP Server Outbound Proxy SIP User ID Authenticate ID Authentication Password Name Use DNS SRV User ID is Phone Number SIP Registration IP address or Domain name provided by VoIP service provider IP address or Domain name o
2. Hz Message Waiting Distinct Ring Tones Gn enna Audible Ringing Busy Tone Reorder Tone Receiver offhook Volume Amplification x 28 HandyTone 386 User Manual Grandstream Networks Inc e FXS Port 2 Page SIP Server sipssipserver2 com ce g Sip mycompany com or IP address ee g proxy myprovider com or IP address if any Sas L user part of an SIP address ss be identical to or different from SIP User ID Password purposely not displayed for security protection Name TT 5 tional e g John Doe Outbound Proxy SIP User ID Authenticate ID Use DNS SRV EN E yes User ID is phone a PT eee E No E Yes SIP Registration No Yes Unregister On R t d Bebook K NO E Yes Register Expiration SS in minutes default 1 hour max 45 days local SIP port DIR default 5062 local RTP port 5008 1024 65535 default 5008 ES ves DTMF Payload Type H Send DTMF HD a D RTP RFC2833 Via SIP INFO Send Flash Event Use random port No Eyes Flash will be sent as a DTMF event 1f set to Yes S Eu R UT Yes Call Forwarding amp Call Waiting Disable are supported locally Use Bell style 3 way Conference ES J if Yes 23 will be disabled 29 HandyTone 386 User Manual Grandstream Networks Inc Offhook Auto Dial es D extension to dial automatically when offhook Proxy Require Disable Call
3. Caller ID Scheme Onhook Voltage Polarity Reversal NTP server Syslog Server IP address or domain name of firmware server IP address or domain name of configuration server Default is blank If configured HandyTone 386 will request the 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 Default is Yes For firmware encryption It should be 32 digit in Hexadecimal Representation End user should keep it blank e Bellcore North America e CID Canada e DTMF Brazil e DTMF Sweden e DTMF Denmark e ETSI DTMEF Finland Sweden s ETSI FSK France Germany Norway Taiwan UK CCA Select the onhook voltage to suit different area or PBX Default is No If set to Yes polarity will be reversed upon call establishment and termination URI or IP address of the NTP Network Time Protocol server which the HT386 will use to synchronize the date time The IP address or URL of syslog server especially useful for ITSP Internet Telephone Service Provider 25 HandyTone 386 User Manual Grandstream Networks Inc Syslog Level Select the ATA to report the log level Default is NONE The level is either one of DEBUG INFO WARNING or ERROR Syslog messages are sent based on the following events product model vers
4. User Manual Handy Tone 386 Analog Telephone Adaptor For Firmware Release Version 1 0 3 44 Grandstream Networks Inc www grandstream com A ERANDSTREAM NETWORKS 6 scien Insevative IP Telephony HandyTone 386 User Manual Grandstream Networks Inc Table of Contents M 16 OL 6 1 5 One ope nee eek ND re em err nme E mm eNO A SES Rae renee mn yD mre 4 2 INSTALLATION gruan E AEAEE EA ETTEN EEANN NEA 5 3 WHAT IS INCLUDED INTHE PACKAGE uinn r ET E AT entacsens 7 S SAEN COMPLIANCE Sraa O E E OO ENEN 7 IA WARRANTY TTT 7 a PRODUCTION ERY UE W ereire a A E E A boca 8 Ak ARB Y NBA TURES TTT 8 AD HARDWARE OPECIFICA TION airaa a E 9 Dy BASICOPERA TION Sres E A O EE 10 Sk GET FAMILIAR WITH VOICE PROMPT erento E EA 10 32 MAKE PHONE CALES cuiii E roeatiocannapeevateieeunaaeieane aes 11 5 2 1 Calling phone or extension numbers esse sees eee 11 I CCE CaS aaae T 11 Dee 91 lt La T a a E TEA EE T ante aaa aise 12 s Cal WAUN ae a a 12 IAr CKS TTT 12 KO Way COMS NCN Derr a A 13 5 2 6 1 Star code style 3 way Conmrerencine nsc 4c overursiadeiniorstaiciansdeciereussan 13 3 210 2 Bellcore style 3 way COnlerenCe ssania e stuns areaseungeed 14 32 FPSIN Pass Thr Ou d Dear a a aa beaaaccege 14 Do UCR FEATURE TTT 14 5 5 1 Gall Peares Table Star Code TT 14 S322 PSTN Pass Through OLITE ET vgs sss cs tales de ents a assed eiateee 15 Oe FAX SUPPOR TTT 15 Sr JEBD LIGHT PATTERN CTET eO 16 0 CONFIGURATION GUE T 1
5. with no specified STUN server the HT386 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 The HT386 supports 6 different codec types including G 711 A Ulaw G 723 1 G 726 G 729A B iLBC A user can configure codecs in a preference list that will be included with the same preference order in SDP message This field contains the number of voice frames to be transmitted in a single packet When setting this value the user should be aware of the requested packet time used in SDP message as a result of configuring this parameter This parameter is associated with the first codec in the above codec Preference List or the actual used payload type negotiated between the 2 conversation parties at run time e g if the first codec is configured as G723 and the Voice Frames per TX is set to be 2 then the ptime value in the SDP message of an INVITE request will be 60ms because each G723 voice frame contains 30ms of audio Similarly if this field is set to be 2 and if the first codec chosen is G729 or G711 or G726 then the ptime value in the SDP message of an INVITE request will be 20ms If the configured voice frames per TX exceeds the maximum allowed value the HT386 will use and save the maximum allowed value for the corresponding first codec choice The maximum value for PCM is 10 x10ms frames for G726 it is 20 xlOms f
6. Features G Yes if Yes Call Forwarding amp Call Waiting Disable are supported locally Use Bell style oe l l l 3 way Conference Yes UT Yes 23 will be disabled Offhook Auto Dial ee I ID extension to dial automatically when offhook Proxy Require Disable Call Waiting NAT Traversal STUN me E O A seconds default is 4 seconds MASE be choice 2 choice 3 choice 4 choice 5 choice 6 Voice Frames per o to 10 20 32 64 for G711 G726 G723 other codecs respectively EE HandyTone 386 User Manual Grandstream Networks Inc Lx G723 rate E 3kbps encoding rate ERs 3100s encoding rate R ins ES s T 96 and 127 default is 97 No 2 iLBC frame size iLBC payload type Silence Suppression Fax Mode C 38 Auto Detect B ush Early Dial oe ES use Yes only if proxy supports 484 response Dial Plan Prefix E prefix string is added to each dialed number Use as Dial Key ES Ea if set to Yes will function as the Re Dial key P L Rua do not send SUBSCRIBE for Message Waiting Indication 7 send periodical SUBSCRIBE for Message Waiting Indication Send Anonymous EAN ves caller ID will be blocked if set to Yes Lock keypad update No 3 configuration update via keypad is disabled 1f set to Yes Special Feature S CD Q O lt ke Frequency2 ON x10ms OFF x10ms Hz C1 C2 C3 C1 C2 C3 Dial Tone Recall Dial Tone
7. appropriate responses in the form of voice fax callback e mail and perhaps other media MTU A Maximum Transmission Unit MTU is the largest size packet or frame specified in octets eight bit bytes that can be sent in a packet or frame based network such as the Internet The maximum for Ethernet is 1500 byte NAT Network Address Translation NTP Network Time Protocol a protocol to exchange and synchronize time over networks The port used is UDP 123 Grandstream products using NTP to get time from Internet 43 HandyTone 386 User Manual Grandstream Networks Inc OBP SBC Outbound Proxy or another name Session Border Controller A device used in VoIP networks OBP SBCs are put into the signaling and media path between calling and called party The OBP SBC acts as if it was the called VoIP phone and places a second call to the called party The effect of this behaviour is that not only the signaling traffic but also the media traffic voice video etc crosses the OBP SBC Without an OBP SBC the media traffic travels directly between the VoIP phones Private OBP SBCs are used along with firewalls to enable VoIP calls to and from a protected enterprise network Public VoIP service providers use OBP SBCs to allow the use of VoIP protocols from private networks with internet connections using NAT PPPoE Point to Point Protocol over Ethernet is a network protocol for encapsulating PPP frames in Ethernet frames It is used
8. e Ifyou CAN NOT log into the configuration page by using default password please check with the VoIP service provider Most likely the VoIP service provider has provisioned the device and configured for you therefore the password has already been changed After a correct password is entered in the login screen the embedded Web server inside the HandyTone 386 will respond with the Configuration pages which are explained in details below e Status Page 18 HandyTone 386 User Manual Grandstream Networks Inc MAC Address 00 0B 82 00 00 00 IP Address 192 168 1 109 Product Model HT386 Software Version Program 1 0 3 44 Bootloader 1 0 8 11 HTML 1 0 3 44 VOC 1 0 0 10 System Up Time 0 day s 0 hour s 2 minute s Registered Yes PPPoE Link Up disabled NAT detected NAT type is full cone MAC Address The device ID in HEX format This is very important ID for ISP troubleshooting IP Address This field shows IP address of the HT386 Product Model This field contains the product model info such as HT386 Software Version Program This is the main software release This number is always used for firmware upgrade Current release is 1 0 3 44 Bootloader current version is 1 0 8 11 HTML current version 1 0 3 44 VOC current version is 1 0 0 10 System Uptime This shows system up time since last reboot Registered Whether the unit is registered to service provider s server PPPoE Link Up This shows whether the PP
9. two FXS ports The RJ 11 jack next to the LAN is called FXS port 2 and the RJ 11 jack on the corner 19 called FXS port 1 The RJ 11 jack on the side on of the HandyTone 386 is a LINE port or PSTN pass through port Each FXS port can have a separate SIP account This is a key feature of HandyTone 386 Both ports can make calls concurrently Following are the steps to install a HandyTone 386 l 3 4 5 Connect a standard touch tone analog telephone or fax machine to FXS port 1 Connect another standard touch tone analog telephone or fax machine to FXS port 2 Insert a standard telephone cable into the LINE port of HandyTone 386 and connect the other end of the telephone cable to a wall jack Insert the Ethernet cable into the LAN port of HandyTone 386 and connect the other end of the Ethernet cable to an uplink port a router or a modem etc Insert the power adapter into the HandyTone 386 and connect it to a wall outlet Please follow the instructions in section 6 2 1 to configure the HandyTone 386 HandyTone 386 User Manual Grandstream Networks Inc 3 What is Included in the Package The HandyTone 386 package contains 1 One HandyTone 386 2 One universal power adaptor 3 One Ethernet cable 3 1 Safety Compliances The HandyTone 386 is compliant with various safety standards including FCC CE and C tick Its power adaptor is compliant with UL standard The HandyTone 386 should only operate with the universal power a
10. 1 LED Green and Red color Universal Switching Input 100 240V AC 50 60 Hz Power Adaptor Output 5VDC 1200mA UL certified Dimension 70mm W 130mm D 27mm H Weight 0 6lbs 0 3kg Temperature 40 130 F 5 45 C Humidity 10 90 non condensing Compliance ce LC HandyTone 386 User Manual Grandstream Networks Inc 5 Basic Operations 5 1 Get Familiar with Voice Prompt HandyTone 386 has a stored voice prompt menu for quick browsing and simple configuration Currently the voice prompt menu and the LED button is designed for FXS port 1 ONLY To enter this voice prompt menu simply press the button or from the analog phone Voice Prompt User s Options Main Menu Enter a Menu Option Enter for the next menu option Enter to return to the main menu Enter 01 06 47 86 or 99 Menu option DHCP Mode Enter 9 to toggle the selection Static IP Mode If user selects Static IP Mode user need configure the all IP address information through menu 02 to 05 If user selects Dynamic IP Mode the device will retrieve all IP address information from DHCP server automatically when user reboots the device IP Address IP address The current WAN IP address is announced Enter 12 digit new IP address if in Static IP Mode O A OS o6 i 4 Direct IP Calling When entered user will be prompted a dial tone dial a 12 digit IP address to make a direct IP
11. 6 1 3 TFTP Server Address Follow section 5 1 with voice menu option 06 to configure the IP address of the TFTP server 6 2 Configuring HandyTone 386 with Web Browser HandyTone 386 ATA 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 HandyTone 386 through a Web browser such as Microsoft s IE AOL s Netscape or Mozilla Firefox 6 2 1 Access the Web Configuration Menu First get the IP address of the HandyTone 386 through section 5 1 with menu option 02 Then access the HandyTone 386 s Web Configuration Menu using the following URI http Phone IP Address where the Phone IP Address is the IP address of the phone NOTE e To type IP address into browser to get the configuration page please strip out the announced leading 0 as the browser will parse in octet e g if the IP address reported 192 168 001 014 please type in 192 168 1 14 6 2 2 End User Configuration 17 HandyTone 386 User Manual Grandstream Networks Inc Once this HTTP request is entered and sent from a Web browser the HandyTone 386 will respond with the following login screen Password is Login 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 administrator can get access to the ADVANCED SETTING configuration page NOTE
12. 7 6 1 CONFIGURING HANDY TONE 386 LAN IP THROUGH VOICE PROMPT ccccceeeeeees 17 Oi ADIGE NOG sis sesso hic tees eats edn ati anata inl einen addons fea taede ene loutnaseet 17 OL 2 K TIC AP MOC TTT 17 CES TET PSS Cry Adde TTT 17 6 2 CONFIGURING HANDYTONE 386 WITH WEB BROWSER cesse eee 17 6 2 1 Access the Web Configuration Menu sese eee eee eee 17 6O22 End User e pE C ee eesi o E 18 6 2 3 Advanced Configuration and FXS ports Parameters cccccccsssseeseeeeeeeeeeees 23 6 2 4 Saving the Configuration Changes sss eee eee 36 6 2 5 Rebooting the HandyTone 386 from Remote esse esse 36 6 3 CONFIGURATION THROUGH A CENTRAL SERVER 37 T SOFTWARE UPGRADE T 38 7 1 FIRMWARE UPGRADE THROUGH TF TP HTTD sse 38 2 HandyTone 386 User Manual Grandstream Networks Inc 7 2 CONFIGURATION FILE DOWNLOAD cccececceccscsccccsccscccscesesceccscesescesescusescesescees 39 7 3 FIRMWARE AND CONFIGURATION FILE PREFIX AND POSTFIX cscescecececcecececess 39 7 4 MANAGING FIRMWARE AND CONFIGURATION FILE DOWNLOAD 0ccccececeecececess 39 8 RESTORE FACTORY DEFAULT SETTING aoaaa nnannnnnnnnennnrnnnn nn 40 9 E RES eN TTT 41 HandyTone 386 User Manual Grandstream Networks Inc 1 Welcome Congratulations on becoming an owner of HandyTone 386 You made an excellent choice and we hope you enjoy all of its capabilities Grandstream s HandyTone 386 is an all in one VoIP integrated access device that featur
13. File download in minutes to reduce the Firmware or Provisioning Server load at any given time 38 HandyTone 386 User Manual Grandstream Networks Inc 8 Restore Factory Default Setting Warning Restore the Factory Default Setting will DELETE all configuration information of the device Please backup or print out all the settings before you approach to following steps Grandstream will not take any responsibility tf you lose all the parameters of setting and cannot connect to your service provider Please disconnect network cable and power cycle the unit before trying to reset the unit to factory default The steps are as follows Step 1 Find the MAC Address of the device It is a 12 digits HEX number located on the bottom of the unit Step 2 Encode the MAC address Please use the following mapping 0 9 0 9 A 22 B 222 Ci 2222 D 33 E 333 F 3333 For example if the MAC address is 000b8200e395 it should be encoded as 0002228200333395 Step 3 To perform factory reset Press or the LED button for voice prompt Enter 99 and get the voice prompt Reset Enter the encoded MAC address of the device Wait for 15 seconds Oo o gt The device will reboot automatically and restore to factory default setting 39 HandyTone 386 User Manual Grandstream Networks Inc 9 Glossary of Terms ADSL Asymmetric Digital Subscriber Line Modems attached to twisted pair copper wiring that trans
14. HandyTone 386 will then display the following screen to confirm that the changes have been saved Your configuration changes have been saved They will take effect on next reboot Reboot Users are recommended to Reboot the HandyTone 386 after seeing the above message 6 2 5 Rebooting the HandyTone 386 from Remote The administrator of the HandyTone 386 can remotely reboot the HT386 by clicking on the Reboot button at the bottom of the configuration page Once done the following screen will be displayed to indicate that rebooting is underway 35 HandyTone 386 User Manual Grandstream Networks Inc The device is rebooting now You may relogin by clicking on the link below in 30 seconds Click to relogin At this point the user can relogin to the HandyTone 386 after waiting for about 30 seconds 6 3 Configuration through a Central Server Grandstream HT386 can be automatically configured from a central provisioning system When HT386 boots up it will send TFTP request to download configuration files there are two configuration files one is cfg txt and the other is cfg000b82xxxxxx where OOOb82xxxxxx is the MAC address of the HT386 The configuration files can be downloaded via TFTP from the central server A service provider or an enterprise with large deployment of HT386 can easily manage the configuration and service provisioning of individual devices remotely from a central server Gran
15. PC running the TFTP server and the HandyTone ATA in the same LAN segment 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 Start the TFTP server in the HandyTone ATA s web configuration page configure the Firmware Server Path with the IP address of the PC update the change and reboot the unit Please be advised that our client will pull out firmware from the WAN side if the TFTP server is connected to the device s LAN port the firmware upgrade will not work by design 3 HandyTone 386 User Manual Grandstream Networks Inc 7 2 Configuration File Download Grandstream SIP Device can be configured via Web Interface as well as via Configuration File through TFTP or HTTP Config Server Path is the TFTP or HTTP 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 19 associated with each particular field in the web configuration page A parameter consists of a Capital letter P and 2 to 3 Could be extended to 4 in the future digit numeric numbers 1 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 Grandstream Device boots u
16. PoE is up if connected to DSL modem NAT This shows what kind NAT the HT386 is connected to Itis based on STUN protocol If the detected NAT is symmetric NAT STUN will not work and Outbound Proxy needed to make HT386 functioning correctly e Basic Settings Page 19 HandyTone 386 User Manual Grandstream Networks Inc End User Password E poscly not displayed for security protection Web Port ET aul for HTTP is 80 IP Address ynamically assigned via DHCP default or PPPoE will attempt PPPoE if DHCP fails and following is non blank DHCP hostname DHCP domain DHCP vendor class ID PPPoE account ID PPPoE password PPPoE Service Name Preferred DNS server o fo E E 7 HT configured as IP Address Subnet Mask Default Router DNS Server 1 DNS Server 2 Time Zone GwT 5 00 USEastem Tine NewYork Daylight pc 7 Savings Time Bu Ea Optional Rule 4 1 7 2 0 10 1 7 2 0 60 PSTN access oode as pattern to use the PSTN line default is 00 20 HandyTone 386 User Manual Grandstream Networks Inc End User Password Web Port IP Address DHCP hostname DHCP domain DHCP vendor class ID Time Zone This contains the password for end user to access the Web Configuration Menu User can put new password here This field is case sensitive with maximum of 25 characters This is the device s internal HTTP server port Default is 80 If DHCP mode is en
17. STN outgoing calls This is called PSTN Pass Through 22 HandyTone 386 User Manual Grandstream Networks Inc 6 2 3 Advanced Configuration and FXS ports Parameters To login to the Advanced Setting and FXS port configuration pages administrator password is required The default administrator password is admin User can change the administrator password here The password is case sensitive and the maximum length is 25 characters e Advanced Settings Page a D not displayed for security protection Home NPA a Layer 3 QoS R DiffServ or Precedence value Admin Password Layer 2 QoS 802 1Q VLAN Tag a 802 1p priority value E STUN server is Stimsipseniereom TT URI or IP port keep alive interval O A seconds default 20 seconds Use NAT IP N in SIP SDP message 1f specified Firmware Upg 6 ade and Upgrade Via E TFTP Ehr TP Provisioning Firmware Server Path Config Server Path Firmware File Prefix Firmware File Postfix Config File Prefix a Config File Postfix Automatic Upgrade ES Ea vcs check for upgrade every 0080 nutes default 7 days ways Check for New Firmware 7 check New Firmware only when F W pre suffix changes 23 HandyTone 386 User Manual Grandstream Networks Inc Firmware Key as I Hexadecimal Representation Onhook Threshold 800m FXS Impedance Caller ID Scheme Onhook Voltage No Ely es reverse polarity upon call establishment and
18. Z o o o Enable Call Waiting Per Call ce Unconditional Call Forward To use this feature dial 72 wait for the dial tone Then dial the forward number ended with wait for dial tone hang up dial tone then hang up N Blind Transfer Refer 5 2 5 1 above for procedure to perform Blind Transfer 70 73 Cancel Unconditional Call Forward To cancel Unconditional Call Forward dial 73 and get the 87 HandyTone 386 User Manual Grandstream Networks Inc Busy Call Forward To use this feature dial 90 wait for the dial tone Then dial the forward number ended with wait for dial tone hang up Cancel Busy Call Forward To cancel Busy Call Forward dial 91 and get the dial tone then hang up Delayed Call Forward To use this feature dial 92 wait for the dial tone Then dial the forward number ended with wait for dial tone hang up Cancel Delayed Call Forward To cancel this Forward dial 93 and get the dial tone then hang up Flash Hook When in conversation this action will switch to the new incoming call if user heard the call waiting sound When in conversation and no incoming call heard this action will switch to a new channel for a new call 5 3 2 PSTN Pass Through Life Line When HandyTone 386 is out of power the RJ 11 line jack on the HandyTone 386 side will function as a pass through jack The user will be able to use the analog phone for PSTN cal
19. abled then all the field values for the Static IP mode are not used even though they are still saved in the Flash memory The HT386 will acquire its IP address from DHCP in the network PPPoE settings is usually for DSL ADSL modem users The HandyTone will attempt to establish a PPPoE session if PPPoE account is set If Static IP mode is selected the IP address Subnet Mask Default Router IP address DNS Server 1 mandatory DNS Server 2 optional fields need to be configured 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 blank This option is used by clients and servers to exchange vendor specific information Default is blank This parameter controls how the displayed date time will be adjusted according to the specified time zone 21 HandyTone 386 User Manual Grandstream Networks Inc Daylight Savings Time This parameter controls whether the displayed time will be daylight savings PSTN Access Code time or not If set to Yes and the Optional Rule is empty then the displayed time will be 1 hour ahead of normal time The Automatic Daylight Saving Time Rule shall have the following syntax start time end time saving Both start time and end time have the same syntax month day weekday h
20. call For details see 4 2 2 Make a Direct IP Call Voice Messages Pending If there are voice messages user can dial 9 No Voice Messages and dial pre configured phone number to retrieve voice message RESET Enter 9 to reboot the device or Enter MAC address to restore factory default U 02 03 04 05 7 setting For details see section 8 Invalid Entry Automatically return to Main Menu 10 HandyTone 386 User Manual Grandstream Networks Inc NOTE 5 2 5 2 1 Once the button is pressed it enters the voice prompt main menu If the button is pressed again while it is already in the voice prompt menu it jumps to Direct IP Call option and a dial tone is prompted shifts down to the next menu option returns to the main menu 9 functions as the ENTER key in many cases to confirm an option 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 like 192 168 0 26 should be key in like 192168000026 no dot needed while input Once all of the digits are collected the input will be processed Key entry can not be deleted but the phone may prompt error once it is detected Make Phone Calls Calling phone or extension numbers There are currently two methods to make an extension number call a Dial the numbers directly and wait f
21. ce 23 will be disabled This parameter allows a user to configure a User ID or extension number to be automatically dialed upon offhook Please note that only the user part of a SIP address needs to be entered here The HT386 will automatically append the and the host portion of the corresponding SIP address NOTE Please write down the IP address of the ATA if you use this feature as it will disable the IVR and the only way to access the HT386 is via web configuration page SIP Extension to notify SIP server that the unit is behind the NAT Firewall Default is No User can use code to use this feature per call basis 32 HandyTone 386 User Manual Grandstream Networks Inc NAT Traversal Preferred Vocoder Voice Frames per TX G723 Rate iLBC frame size iLBC payload type This parameter defines whether the HT386 NAT traversal mechanism will be activated or not If activated by choosing Yes and a STUN server is also specified then the HT386 will behave according to the STUN client specification Under this mode the embedded STUN client inside the HT386 will attempt to detect if and what type of firewall NAT it is sitting behind through communication with the specified STUN server If the detected NAT is a Full Cone Restricted Cone or a Port Restricted Cone the HT386 will attempt to 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
22. daptor provided in the package 3 2 Warranty Grandstream has a reseller agreement with our reseller customer End users should contact the company from whom you purchased the product 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 Grandstream reserves the right to remedy warranty policy without prior notification Warning Please do not attempt to use a different power adaptor Using other power adaptor may damage the HandyTone 386 and will void the manufacturer warranty 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 Information in this document is subject to change without notice No part of this document may be reproduced or transmitted in any form or by any means electronic or mechanical for any purpose without the express written permission of Grandstream Networks Inc HandyTone 386 User Manual Grandstream Networks Inc 4 1 4 Product Overview Key Features Supports SIP 2 0 RFC 3261 TCP UDP IP RTP RTCP HTTP ICMP ARP RARP DNS DHCP both client and server NTP PPPoE STUN TFTP etc Supports dual SIP accounts via dual FXS ports Powerful digital signal processing DSP to ensure superb audio quali
23. dstream provides a licensed provisioning package called GAPSLite that can be used to support automated configuration of HandyTone ATA GAPSLite is a light version of GAPS Grandstream Automated Provisioning System that uses enhanced NAT friendly TFTP and SIP protocol to communicate with each individual HandyTone ATA for firmware upgrade remote reboot etc For details on how GAPSLite works please refer to the documentation of GAPSLite product 36 HandyTone 386 User Manual Grandstream Networks Inc 7 Software Upgrade Software upgrade can be done via either TFTP or HTTP The corresponding configuration settings are in the ADVANCED SETTINGS configuration page 7 1 Firmware Upgrade through TFTP HTTP To upgrade via TFTP or HTTP the Firmware Upgrade and Provisioning upgrade via field needs to be set to TFTP or HTTP 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 3 44 e g 168 75 215 190 NOTES e TFTP server in IP address format can be configured via IVR Please refer to section 6 1 3 for instructions If TFTP server is in FQDN format it must be set via web configuration interface 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
24. es superb audio quality rich functionalities high level of integration compactness and ultra affordability The HandyTone 386 is fully compatible with SIP industry standard and can interoperate with many other SIP compliant devices and software on the market Grandstream HandyTone 386 is another addition to the popular HandyTone product family The HandyTone 386 features two FXS ports each with independent SIP accounts This document is subject to changes without notice The latest electronic version of this user manual can be downloaded from the following location http www grandstream com y downloads htm HandyTone 386 User Manual Grandstream Networks Inc 2 Installation HandyTone 386 Analog Telephone Adaptor is an all in one VoIP integrated device designed to be a total solution for networks providing VoIP services The HandyTone 386 VoIP functionalities are available via regular analog telephones The following photo illustrates the appearance of a HandyTone 386 Top View Side Views RJ 11 Line Port a Line 9V 1 2A va RJ 45 10M RJ 11 RJ 11 Ethernet LAN Port BUTTON FXS Port 1 FXS Port 2 RED LED GREEN LED Interconnection Diagram of the HandyTone 386 2 1 Internet ADSL Cable Modem or Ethernet Analog Phone Analog Phone 2 2 LAN FXS 2 FXS 1 Cordless Phone Cordless Phone LINE Le PSTN Line pee Fax Fax HandyTone 386 User Manual Grandstream Networks Inc HandyTone 386 has
25. ess flash back to talk to B 5 2 6 2 Bellcore Style 3 way Conference 13 HandyTone 386 User Manual Grandstream Networks Inc Bellcore style 3 way conference is also supported To do this user needs to enable Use Bell style 3 way Conference in FXS1 or FXS2 web configuration Assuming that call party A and B are in conversation A wants to bring C in a 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 3 If C answers the call then A press flash to bring B C in the conference 4 If C does not answer the call A can press flash back to talk to B 5 2 7 PSTN Pass Through HandyTone 386 supports PSTN pass through on FXS port 1 User can make and receive PSTN calls with attached analog phone in Phone 1 port Phone 2 port or FXS port 2 does NOT have this feature e To receive PSTN calls simply make phone off hook when the analog phone rings e To make a PSTN call simply press the PSTN access code 00 is default or any number configured in web configuration page to switch to the PSTN line and get dial tone then dial the number 5 3 Call Features 5 3 1 Call Features Table Star Code Following table shows the call features code of HandyTone 386 Call Features t23 3 way Conferencing Refer 5 2 6 above for procedure to perform 3 way Calling 70 Disable Call Waiting Per Cal Z o
26. f Outbound Proxy or Media Gateway or Session Border Controller Used by ATA for firewall or NAT penetration in different network environment If symmetric NAT is detected STUN will not work and ONLY Outbound Proxy will work User account information provided by VoIP service provider ITSP usually has the form of digit similar to phone number or actually a phone number ID used for authentication usually same as SIP user ID but could be different and decided by ITSP Password for ATA to register to SIP servers of ITSP Purposely blank out once saved for security Maximum length is 25 User name not user ID for information only Default is No If set to Yes the client will use DNS SRV to lookup for the server If the HandyTone If set to yes a user phone parameter will be attached to the From header in SIP request This parameter controls whether the HT386 needs to send REGISTER messages to the proxy server The default setting is Yes 31 HandyTone 386 User Manual Unregister On Reboot Register Expiration Local SIP port Local RTP port Use Random Port DTMF Payload Type Send DTMF Send Flash Event Enable Call Features Use Bell style 3 way Conference Offhook Auto Dial Proxy Require Disable Call Waiting Grandstream Networks Inc Default is No If set to Yes the device will first send registration request to indicate SIP registra to remove previous bindings This pa
27. image is available the HandyTone ATA will attempt to retrieve the new image files by downloading them into the HandyTone ATA s SRAM During this stage the HandyTone ATA 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 fails for any reason e g TFTP HTTP server is not responding there are no code image files available for upgrade or checksum test fails etc the HandyTone ATA will stop the TFTP HTTP process and simply boot using the existing code image in the flash e Firmware upgrade may take as long as 1 to 20 minutes over Internet or just 20 seconds 1f it is performed on a LAN It 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 Please check the Services section of Grandstream s Web site to obtain our public TFTP server s IP address e Alternatively 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 y firmware htm Unzip the file and put all of them under the root directory of the TFTP server Put the
28. ion 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 memory exception ERROR 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 Here is an example May 19 02 40 38 192 168 1 14 GS_LOG 00 0b 82 00 a1 be 000 Ethernet link is up e FXS Port 1 Page SIP Server Sisipserverl com e g sip mycompany com or IP address Outbound Proxy O OA g proxy myprovider com or IP address if any SIP User ID Ss user part of an SIP address Authenticate ID E be identical to or different from SIP User ID purposely not displayed for security protection Password Name IR aal e g John Doe Use DNS SRV Ey ES 26 HandyTone 386 User Manual Grandstream Networks Inc User ID is phone number SIP Registration Unregister On Reboot Register Expiration OT in minutes default 1 hour max 45 days local SIP port 500 default 5060 local RTP port 5008 1024 65535 default 5004 Use random port E No E yes DTMF Payload P Type H Send DTMF WE audio i H RTP RFC2833 _ Y SIP INFO Send Flash Event E S Enable Call
29. ll if the time is not short enough 5 2 4 Call Waiting If call waiting feature is enabled while the user is in a conversation he will hear a special stutter tone if there is another incoming call User can press the flash button to put the current call party on hold and switch to the other call Pressing flash button toggles between two active calls 5 2 5 Call Transfer 5 2 5 1 Blind Transfer Assume that call party A and B are in conversation A wants to Blind Transfer B to C 1 A press FLASH on the analog phone to hear the dial tone 2 Then A dials 87 then dials C s number and then or wait for 4 seconds 3 Acan hang up NOTE 12 HandyTone 386 User Manual Grandstream Networks Inc Enable Call Feature has to be set to Yes in web configuration page A can hold on to the phone and await one of the three following behaviors A quick confirmation tone temporarily using the call waiting indication tone followed by a dial tone This indicates the transfer is successful transferee has received a 200 OK from transfer target At this point A can either hang up or make another call 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 Busy tone keeps playing This means we have failed to receive the sec
30. ls directly without press the access code 5 4 FAX Support HandyTone 386 supports FAX in two modes T 38 Fax over IP and 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 may be used To send or receive faxes in fax pass through mode users must select all the Preferred Codecs to be PCMU PCMA G 71 1 u a 5 5 LED Light Pattern Indication Following are the LED light pattern indications RED LED always indicates not abnormal status DHCP Failed or WAN No Cable Button flashes every 2 seconds if DHCP is configured 15 HandyTone 386 User Manual Grandstream Networks Inc HandyTone 486 fails to register Button flashes every 2 seconds if SIP server is configured Firmware Upgrading Button flashes every 2 seconds Device Malfunctions Red light steady on 16 HandyTone 386 User Manual Grandstream Networks Inc 6 Configuration Guide 6 1 Configuring HandyTone 386 LAN IP through Voice Prompt 6 1 1 DHCP Mode Follow section 5 1 with voice menu option 01 to enable HandyTone 386 to use DHCP 6 1 2 STATIC IP Mode Follow section 5 1 with voice menu option 01 to enable HandyTone 386 to use STATIC IP mode then use option 02 03 04 to set up HandyTone 386 s IP Subnet Mask Gateway respectively
31. luded as part of the dial string to be sent out Default is NO When set to Yes a SUBSCRIBE for Message Waiting Indication will be sent periodically If this parameter is set to Yes user ID will be sent as anonymous essentially block the Caller ID from displaying If set to Yes the configuration update via keypad is disabled NOTE Since only FXSI has LED for indication and IVR for keypad access this field is not applied to FXS2 Default is Standard Choose the selection to meet some special requirements from Soft Switch vendors like Lucent Nortel BroadSoft etc Using these settings user can configure ring or tone frequencies according to their preference By default they are set to North American frequencies Frequencies should be configured 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 ring OFF should be zero Otherwise it will ring ON ms and a pause of OFF ms and then repeat the pattern Handset volume adjustment RX is for receiving volume TX is for transmission volume Default values are OdB for both parameters 6dB generates the highest volume and 6dB generates the lowest volume 34 HandyTone 386 User Manual Grandstream Networks Inc 6 2 4 Saving the Configuration Changes Once a change is made users should click on the Update button in the Configuration page The
32. mainly with cable modem and DSL services PSTN Public Switched Telephone Network i e the phone service we use for every ordinary phone call or called POT Plain Old Telephone or circuit switched network RTCP Real time Transport Control Protocol defined in RFC 3550 a sister protocol of the Real time Transport Protocol RTP It partners RTP in the delivery and packaging of multimedia data but does not transport any data itself It is used periodically to transmit control packets to participants in a streaming multimedia session The primary function of RTCP is to provide feedback on the quality of service being provided by RTP RTP Real time Transport Protocol defines a standardized packet format for delivering audio and video over the Internet It was developed by the Audio Video Transport Working Group of the IETF and first published in 1996 as RFC 1889 SDP Session Description Protocol is a format for describing streaming media initialization parameters It has been published by the IETF as RFC 2327 SIP Session Initiation Protocol An IP telephony signaling protocol developed by the IETF RFC3261 SIP is a text based protocol suitable for integrated voice data applications SIP is 44 HandyTone 386 User Manual Grandstream Networks Inc designed for voice transmission and uses fewer resources and 1s considerably less complex than H 323 All Grandstream products are SIP based STUN Simple Traversal of UDP over N ATS i
33. mit from 1 5 Mbps to 9 Mbps downstream to the subscriber and from 16 kbps to 800 kbps upstream depending on line distance AGC Automatic Gain Control is an electronic system found in many types of devices Its purpose is to control the gain of a system in order to maintain some measure of performance over a changing range of real world conditions ARP Address Resolution Protocol is a protocol used by the Internet Protocol IP RFC826 pecifically IPv4 to map IP network addresses to the hardware addresses used by a data link protocol The protocol operates below the network layer as a part of the interface between the OSI network and OSI link layer It is used when Pv4 is used over Ethernet ATA Analogue Telephone Adapter Covert analogue telephone to be used in data network for VoIP like Grandstream HT series products CODEC Abbreviation for Coder Decoder It s an analog to digital A D and digital to analog D A converter for translating the signals from the outside world to digital and back again CNG Comfort Noise Generator geneate artificial background noise used in radio and wireless communications to fill the silent time in a transmission resulting from voice activity detection DATAGRAM A data packet carrying its own address information so it can be independently routed from its source to the destination computer DECIMATE To discard portions of a signal in order to reduce the amount of information to be encoded o
34. n FXS device will allow any FXO device to operate as if it were connected to the phone company This makes your PBX the POTS PSTN for the phone The FXS Interface connects to FXO devices by an FXO interface of course DHCP The Dynamic Host Configuration Protocol DHCP is an Internet protocol for automating the configuration of computers that use TCP IP DHCP can be used to automatically assign IP addresses to deliver TCP IP stack configuration parameters such as the subnet mask and default router and to provide other configuration information such as the addresses for printer time and news servers ECHO CANCELLATION Echo Cancellation is used in telephony to describe the process of removing echo from a voice communication in order to improve voice quality on a telephone call In addition to improving quality this process improves bandwidth savings achieved through silence suppression by preventing echo from traveling across a network There are two types of echo of relevance in telephony acoustic echo and hybrid echo Speech compression techniques and digital processing delay often contribute to echo generation in telephone networks 42 HandyTone 386 User Manual Grandstream Networks Inc H 323 A suite of standards for multimedia conferences on traditional packet switched networks HTTP Hyper Text Transfer Protocol the World Wide Web protocol that performs the request and retrieve functions of a server IP Internet Protoc
35. ol A packet based protocol for delivering data across networks IP PBX IP based Private Branch Exchange IP Telephony Internet Protocol telephony also known as Voice over IP Telephony A general term for the technologies that use the Internet Protocol s packet switched connections to exchange voice fax and other forms of information that have traditionally been carried over the dedicated circuit switched connections of the public switched telephone network PSTN The basic steps involved in originating an IP Telephony call are conversion of the analog voice signal to digital format and compression translation of the signal into Internet protocol IP packets for transmission over the Internet or other packet switched networks the process is reversed at the receiving end The terms IP Telephony and Internet Telephony are often used to mean the same however they are not 100 per cent interchangeable since Internet is only a subcase of packet switched networks For users who have free or fixed price Internet access IP Telephony software essentially provides free telephone calls anywhere in the world However the challenge of IP Telephony is maintaining the quality of service expected by subscribers Session border controllers resolve this issue by providing quality assurance comparable to legacy telephone systems IVR IVR is a software application that accepts a combination of voice telephone input and touch tone keypad selection and provides
36. ond NOTIFY from the transferee and decided to time out Note this does not indicate the transfer has been successful nor does it indicate the transfer has failed When transferee is a client that does not support the second NOTIFY such as our own earlier firmware this will be the case In bad network scenarios this could also happen although the transfer may have been completed successfully 5 2 5 2 Attended Transfer Assume that call party A and B are in conversation A wants to Attend Transfer B to C 1 A presses FLASH on the analog phone to get a dial tone 2 A then dial C s number followed by or wait for 4 seconds 3 If C answers the call A and C are in conversation Then A can hang up to complete transfer 4 If C does not answer the call A can press flash back to talk to B NOTE When Attended Transfer failed and A hang up the HandyTone 386 will ring user A back again to remind A that B is still on the call A can pick up the phone to restore conversation with B 5 2 6 3 way Conferencing 5 2 6 1 Star Code Style 3 way Conference Assuming that call party A and B are in conversation A wants to bring C in a conference l 2 3 4 A presses FLASH on the analog phone or Hook Flash for old model phones to get a dial tone A dials 23 then C s number then or wait for 4 seconds If C answers the call then A press flash to bring B C in the conference If C does not answer the call A can pr
37. or 4 default seconds b Dial the numbers directly and press assuming that use as dial key is selected in web configuration Examples To dial another extension on the same proxy such as 1008 simply pick up the attached phone dial 1008 and then press the or wait for 4 seconds To dial a PSTN number such as 6266667890 you might need to enter in some prefix number followed by the phone number Please check with your VoIP service provider to get the information If you phone is assigned with a PSTN like number such as 6265556789 most likely you just follow the rule to dial 16266667890 as if you were calling from a regular analog phone of North America then followed by pressing or wait for 4 seconds 5 2 2 Direct IP calls Direct IP calling allows two parties that is a HandyTone with an analog phone and another VoIP Device to talk to each other in an ad hoc fashion without a SIP proxy This kind of VoIP calls can be made between two parties if Both HT386 and other VoIP Device i e another HandyTone ATA or Budgetone SIP phone or other VoIP unit have public IP addresses or Both HT386 and other VoIP Device are on the same LAN using private IP addresses or Both HT386 and other VoIP Device can be connected through a router using public or private IP addresses with necessary port forwarding or DMZ 11 HandyTone 386 User Manual Grandstream Networks Inc To make a direct IP call first pick up the analog phone or tu
38. our minute month 1 2 3 12 for Jan Feb Dec day l 1 2 3 31 weekday 1 2 3 7 for Mon Tue Sun or O which means the daylight saving rule is not based on week days but based on the day of the month hour hour 0 23 minute minute 0 59 If weekday is 0 it means the date to start or end daylight saving is at exactly the given date In that case the day value must not be negative If weekday is not zero and day is positive then the daylight saving starts on the first day th iteration of the weekday Ist Sunday 3rd Tuesday etc If weekday us not zero and day is negative then the daylight saving starts on the last day th iteration of the weekday last Sunday 3rd last Tuesday etc The saving is in the unit of minutes The saving time may also be preceded by a negative sign if subtraction is desired instead of addition The default value for Automatic Daylight Saving Time Rule shall be set to 04 01 7 02 00 10 1 7 02 00 60 which is the rule for US Examples US Canada where daylight saving time is applicable 04 01 7 02 00 10 1 7 02 00 60 This means the daylight saving time starts from the first Sunday of April at 2AM and ends the last Sunday of October at 2AM The saving is 60 minutes 1hour Default is 00 user can change it By pressing the code user can switch the phone to PSTN line connected to the Line port of ATA and make P
39. p or reboots it will issue request for configuration file named CLEXXXXXXXXXXXX Where XXXXXXXXXxxx is the MAC address of the device Le cfg000b820102ab The configuration file name should be in lower cases 7 3 Firmware and Configuration File Prefix and Postfix Firmware Prefix and Postfix allows device to download the firmware name with the matching Prefix and Postfix This makes it the possible to store ALL of the firmware with different version in one single directory Similarly Config File Prefix and Postfix allows device to download the configuration file with the matching Prefix and Postfix Thus multiple configuration files for the same device can be stored in one directory In addition when the field Check New Firmware only when F W pre suffix changes is set to Yes the device will only issue firmware upgrade request if there are changes in the firmware Prefix or Postfix 7 4 Managing Firmware and Configuration File Download When Automatic Upgrade is set to Yes Service Provider can use P193 Auto Check Interval in minutes default and minimum is 60 minutes to have the devices periodically check with either Firmware Server or Config Server whenever they are defined 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
40. r compressed Lossy compression algorithms ordinarily decimate while subsampling DECT Digital Enhanced Cordless Telecommunications A standard developed by the European Telecommunication Standard Institute from 1988 governing pan European digital mobile 40 HandyTone 386 User Manual Grandstream Networks Inc telephony DECT covers wireless PBXs telepoint residential cordless telephones wireless access to the public switched telephone network Closed User Groups CUGs Local Area Networks and wireless local loop The DECT Common Interface radio standard is a multicarrier time division multiple access time division duplex MC TDMA TDD radio transmission technique using ten radio frequency channels from 1880 to 1930 MHz each divided into 24 time slots of 10ms and twelve full duplex accesses per carrier for a total of 120 possible combinations A DECT base station an RFP Radio Fixed Part can transmit all 12 possible accesses time slots simultaneously by using different frequencies or using only one frequency All signaling information is transmitted from the RFP within a multiframe 16 frames Voice signals are digitally encoded into a 32 kbit s signal using Adaptive Differential Pulse Code Modulation DNS Short for Domain Name System or Service or Server an Internet service that translates domain names into IP addresses DID Direct Inward Dialing Direct Inward Dialing The ability for an outside caller to dial to a PBX ex
41. rames for G723 it is 32 x30ms frames for G729 G728 64 x10ms and 64 x2 5ms frames respectively Please be careful when massage those parameters Encoding rate for G723 codec By default 6 3kbps rate is set iLBC packet frame size Default is 20ms For Asterisk PBX 30ms might need to be set Payload type for iLBC Default value is 97 The valid range is between 96 and 127 33 HandyTone 386 User Manual Grandstream Networks Inc Silence Suppression Fax Mode Early Dial Dial Plan Prefix Use as Dial Send Key Subscribe for MWI Send Anonymous Lock keypad update Special Feature Distinct Ringtones Volume Amplification This controls the silence suppression VAD feature of G723 and G729 If set to Yes when a silence is detected small quantity of VAD packets instead of audio packets will be sent during the period of no talking If set to No this feature is disabled T 38 Auto Detect FoIP by default or Pass Through must use codec PCMU PCMA Default is No Use only if proxy supports 484 response Sets the prefix added to each dialed number This parameter allows the user to configure the key to be used as the Send or Dial key Once set to Yes pressing this key will immediately trigger the sending of dialed string collected so far In this case this key 19 essentially equivalent to the Re Dial key If set to No this key will then be inc
42. rameter allows the user to specify the time frequency in minutes the HT386 will refresh its registration with the specified registrar The default interval is 60 minutes or 1 hour The maximum interval is 65535 minutes about 45 days This parameter defines the local SIP port the HT386 will listen and transmit The default value is for FXS1 is 5060 FXS2 is 5062 This parameter defines the local RTP RTCP port pair the HT386 will listen and transmit It is the base RTP port for channel 0 When configured channel 0 will use this port_value for RTP and the port_value 1 for its RTCP channel 1 will use port_value 2 for RTP and port_value 3 for its RTCP The default value for FXS1 is 5004 FXS2 is 5008 Default No If set to Yes the device will pick randomly generated SIP and RTP ports This is usually necessary when multiple SIP devices are behind the same NAT For Direct IP to IP call this should be set to No This parameter sets the payload type for DTMF using RFC2833 This parameter specifies the mechanism to transmit DTMF digit There are 3 modes supported in audio which means DTME is combined in audio signal not very reliable with low bit rate codec via RTP RFC2833 or via SIP INFO Default is NO If set to yes flash will be sent as DTMF event Default is Yes Advance call features and feature codes functions are supported locally If this parameter is set to Yes user will be able to make Bellcore style 3 way conferen
43. rn on the speakerphone on the analog phone then access the voice menu prompt by dial or press the button on the HandyTone 286 and dials 47 to access the direct IP call menu User will hear a voice prompt Direct IP Calling and a dial tone Enter a 12 digit target IP address to make a call Destination ports can be specified by 66 99 using 4 encoding for followed by the port number Examples If the target IP address is 192 168 0 10 the dialing convention is Voice Prompt with option 47 then 192 168 000 010 followed by pressing the key if it is configured as a send key or wait for more than 5 seconds If the target IP address port is 192 168 1 20 5062 then the dialing convention would be Voice Prompt with option 47 then 192168001020 45062 followed by pressing the key if it is configured as a send key or wait for 4 seconds NOTE e When doing direct IP call the Use Random Port should set to NO e You can NOT make direct IP calls between FXSI to FXS2 since they are using same IP 5 2 3 Call Hold While in conversation pressing the flash button on the attached analogue phone if the phone has that button will put the remote end on hold Pressing the flash button again will release the previously held party and the bi directional media will resume If no flash button then on off hook quickly hook flash will do the same thing but also risk of losing ca
44. s a network protocol allowing clients behind NAT or multiple NATs to find out its public address the type of NAT it is behind and the internet side port associated by the NAT with a particular local port This information is used to set up UDP communication between two hosts that are both behind NAT routers The protocol is defined in REC 3489 STUN will usually work good with non symmetric NAT routers TCP Transmission Control Protocol is one of the core protocols of the Internet protocol suite Using TCP applications on networked hosts can create connections to one another over which they can exchange data or packets The protocol guarantees reliable and in order delivery of sender to receiver data TFTP Trivial File Transfer Protocol is a very simple file transfer protocol with the functionality of a very basic form of FTP It uses UDP port 69 as its transport protocol UDP User Datagram Protocol UDP is one of the core protocols of the Internet protocol suite Using UDP programs on networked computers can send short messages known as datagrams to one another UDP does not provide the reliability and ordering guarantees that TCP does datagrams may arrive out of order or go missing without notice However as a result UDP is faster and more efficient for many lightweight or time sensitive purposes VAD Voice Activity Detection or Voice Activity Detector is an algorithm used in speech processing wherein the presence or absence of h
45. tension without going through an attendant or auto attendant DSP Digital Signal Processing Using computers to process signals such as sound video and other analog signals which have been converted to digital form Digital Signal Processor A specialized CPU used for digital signal processing Grandstream products all have DSP chips built inside DTMF Dual Tone Multi Frequency The standard tone pairs used on telephone terminals for dialing using in band signaling The standards define 16 tone pairs 0 9 and A F although most terminals support only 12 of them 0 9 and FQDN Fully Qualified Domain Name A FQDN consists of a host and domain name including top level domain For example www grandstream com is a fully qualified domain name www is the host grandstream is the second level domain and com is the top level domain 41 HandyTone 386 User Manual Grandstream Networks Inc FXO Foreign eXchange Office An FXO device can be an analog phone answering machine fax or anything that handles a call from the telephone company like AT amp T They should also operate the same way when connected to an FXS interface An FXO interface will accept calls from FXS or PSTN interfaces All countries and regions have their own standards FXO 19 complimentary to FXS and the PSTN FXS Foreign eXchange Station An FXS device has hardware to generate the ring signal to the FXO extension usually an analog phone A
46. termination NTP Server timenist gov UR or IP address Syslog Server J Syslog Level NONE lt Polarity Reversal Admin Password Administrator password Only administrator can configure the Advanced Settings page Password field is purposely blanked for security reason after clicking update and saved The maximum password length is 25 characters Home NPA Local area code for North American Dial Plan Layer 3 QoS 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 Layer 2 QoS Layer 2 QoS settings Default setting is blank Other VLAN supported equipments required if configured these settings No Key Entry Default is 4 seconds User can short or extend that depends on digits dialed timeout STUN Server IP address or Domain name of the STUN server Keep alive interval Default is 20 seconds The interval of sending dummy UDP packet to keep NAT pin hole open Use NAT IP NAT IP address used in SIP SDP message Default is blank Firmware Upgrade Default method is HTTP Firmware upgrade may take up to 10 minutes and Provisioning depending on network environment Do not interrupt the firmware upgrading process 24 HandyTone 386 User Manual Grandstream Networks Inc Firmware Server Path Config Server Path Firmware File Prefix Firmware File Postfix Config File Prefix Config File Postfix Automatic Upgrade Firmware Key
47. ty advanced adaptive jitter control and packet loss concealment technology Support various codecs including G 711 PCM a law and u law G 723 1 5 3K 6 3K G 726 32K as well as G 729A and iLBC Support Caller D name display or block Call waiting caller ID Hold Call Waiting Flash Call Transfer Call Forward 3 way conferencing in band and out of band DTMF etc Support fax pass through for PCMU and PCMA and T 38 FoIP Fax over IP Support syslog Support Silence Suppression VAD Voice Activity Detection CNG Comfort Noise Generation Line Echo Cancellation G 168 and AGC Automatic Gain Control Support standard encryption and authentication DIGEST using MD5 and MD5 sess Support for Layer 2 802 1Q VLAN 802 1p and Layer 3 QoS ToS DiffServ MPLS Support automated NAT traversal without manual manipulation of firewall NAT Support device configuration via built in IVR Web browser or Central configuration files through TFTP or HTTP server Support firmware upgrade via TFTP or HTTP Support PSTN pass through Ultra compact wallet size and lightweight design great companion for travelers Compact lightweight Universal Power adapter HandyTone 386 User Manual Grandstream Networks Inc 4 2 Hardware Specification The table below lists the hardware specification of HandyTone 386 Model HandyTone 386 LAN interface IxRJ45 10Base T FXS telephone port 2x FXS PSTN Port 1x PSTN pass through or life line port Button
48. uman speech is detected from the audio samples VLAN A virtual LAN known as a VLAN is a logically independent network Several VLANSs can co exist on a single physical switch It is usually refer to the IEEE 802 1Q tagging protocol VoIP Voice over IP VoIP encompasses many protocols All the protocols do some form of signalling of call capabilities and transport of voice data from one point to another e g SIP H 323 etc 45

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