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Grandstream Networks 286 Telephone User Manual

1. ccceccsceccececcsceccsceccecescecess 16 OLL DOE MA e ae E a de 16 OL SAC TAM e De ee tod ee 16 OAS IFIP Serner Addres Soro ed ETO iis 16 6 2 CONFIGURING HANDY TONE 286 WITH WEB BROWSER cccscoscecccceccsceccsceccececesceccscescscescscess 16 6 2 1 Access the Web Configuration Menu ccccccccccccccssssseeesssceecescccsscaaaaaessssseseeseeceessaaaeusseess 16 6227 End User Conf iurati On A A SA A 177 6 2 3 Advanced User CON TOUT ANION 15 A AA AA mentees 22 6 2 4 Saving the Configuration Changes A Ao as 33 6 2 5 Rebooting the HandyTone 286 from Remote oooooonnnccnnnoninnnnnnononnnnonananononnnnnnnnnanoconnnnnnnnos 33 6 3 CONFIGURATION THROUGH A CENTRAL SERVER cscescsceccecsceccccecccceccsceccscesescecescesescusescsencess 34 7 SOFTWARE UPGRADE WITH TETP 2 tdt 35 7 1 FIRMWARE UPGRADE THROUGH TFTP HTTP ccccecececcceccececececececececececcesesecesesacananas 35 17 2 CONFIGURATION FILE DOWNLOAD pron errana L 24 V da da 36 7 3 FIRMWARE AND CONFIGURATION FILE PREFIX AND POSTFIX cccscesceccsceccececcsceccscescscescecess 36 2 Handy Tone 286 User Manual Grandstream Networks Inc 7 4 MANAGING FIRMWARE AND CONFIGURATION FILE DOWNLOAD cccececcecececccececseccecceeeecs 36 8 RESTORE FACTORY DEFAULT SETTING 2 cecccsccccccccccccccccccccccccccccccccccccees 37 9 GLOSSARY OF TERMS sii io 388 Handy Tone 286 User Manual Grandstream Networks Inc
2. Select the ATA 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 e product model version on boot up INFO level e NAT related info INFO level e sent or received SIP message DEBUG level e SIP message summary INFO level e inbound and outbound calls INFO level e registration status change INFO level e negotiated codec INFO level e Ethernet link up INFO level e SLIC chip exception WARNING and ERROR levels e 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 Grandstream implemented SIP Session Timer The session timer extension enables SIP sessions to be periodically refreshed via a re INVITE request Once the session interval expires if there is no refresh via a re INVITE message the session will be terminated Session Expiration is the time in seconds at which the session is considered timed out if no successful session refresh transaction occurs beforehand The default value is 180 seconds The minimum session expiration in seconds The default value is 90 seconds If selecting Yes the phone will use session timer when it makes outbound calls if remo
3. DNS Server IP address Same as Menu option 02 Preferred Vocoder Enter 9 to go to the next selection in the list PCM U PCM A G 723 G 729 iLBC G 726 WAN Port Web Access Enter 9 to toggle between enable disable The current Firmware Server IP address is announced Enter 12 digit new IP address The current Config Server Path IP address is Address announced Enter 12 digit new IP address Upgrade Protocol Upgrade protocol for firmware and configuration update Enter 9 to toggle between NO ay pd NO 13 14 15 TFTP HTTP Handy Tone 286 User Manual Grandstream Networks Inc Firmware Version Firmware version information Firmware Upgrade Firmware upgrade mode Enter 9 to rotate among the following three options always check check when pre suffix changes never upgrade Direct IP Calling When entered user will be prompted a dial tone dial a 12 digit IP address to make a direct IP call For details see 4 2 2 Make a Direct IP Call RESET Enter 9 to reboot the device or Enter MAC address to restore factory default setting For details see section 8 SY Invalid Entry Automatically returns to Main Menu NOTES e Once the LED button is pressed it enters voice prompt main menu If the button is pressed again while it is already in the voice prompt menu state it jumps to Direct IP Calling opt
4. termination NTP Server en URI or ies Send Anonymous Polarity Reversal Anonymous Method E Use From Header Time to ring Special Feature Syslog Server Syslog Level Session Expiration OT in seconds default 180 seconds Min SE A in seconds default and minimum 90 seconds Caller Request Timer O No Request for timer when making outbound calls 24 Handy Tone 286 User Manual Grandstream Networks Inc Callee Request Timer No When caller supports timer but did not request one Force Timer Yes 4 No Use timer even when remote party does not support UAC Specify Refresher UAC E UAS Omit Recommended UAS Specify Refresher UAC E uas When UAC did not specify refresher tag Force INVITE Yes G No Always refresh with INVITE instead of UPDATE Firmware Upgrade and Upgrade Via fmagrandstreamcomgs gt gt Firmware Server Path Config Server Path Firmware File Prefix Firmware File Postfix Fa Config File Prefix Config File Postfix Automatic Upgrade L Yes check for upgrade every e nutes default 7 a Always Skip the Firmware Check ES 1f set to Yes in Hexadecimal Representation J d Yes cfg file would be authenticated before acceptance Authenticate Conf File Lock keypad update Allow conf SIP Account in Basic Settings Override M
5. 1 Welcome Congratulations on becoming an owner of HandyTone 286 You made an excellent choice and we hope you will enjoy all its capabilities Grandstream s award wining HandyTone 286 is innovative Analog Telephone Adaptor that offers a rich set of functionality and superb sound quality at ultra affordable price They are fully compatible with SIP industry standard and can interoperate with many other SIP compliant devices and software on the market 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 user_manuals HandyTone pdtf Handy Tone 286 User Manual Grandstream Networks Inc 2 Installation HandyTone 286 is a VoIP Analog Telephone Adaptor designed to work with an ordinary analog telephone The following photo illustrates the appearance of a HandyTone 286 BUTTON RED LED GREEN LED C L T RJ45 Telephone 10M Ethernet 5V 1200mA Interconnection Diagram of the HandyTone 286 Analog Phone Internet ADSL Cable Modem Ethernet PHONE Cordless Phone Handy Tone 286 User Manual Grandstream Networks Inc 3 What is Included in the Package The HandyTone 286 package contains 1 One HandyTone 286 2 One universal power adaptor 3 One Ethernet cable 3 1 Safety Compliances The HandyTone 286 is compliant with various safety standards including FCC CE and C Tick Its
6. Inc e When Attended Transfer failed if A hangs up the HandTone 286 will ring user A 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 HandyTone 286 supports 3 way conference in two modes which is configurable on the web configuration 5 2 6 1 Bell style conference Assuming that call party A and B are in conversation A wants to bring C in a conference 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 es 5 2 6 2 Non bell style conference Use 23 as conference code to initiate conference Assuming that call party A and B are in conversation A wants to bring C in a conference A presses FLASH on the analog phone or Hook Flash for old model phones to get a dial tone A dials 23 followed by 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 a 5 3 Call Features Following table shows the call features of HandyTone 286 Block Caller ID per call Send Caller ID per call Disable Call Waiting for all subsequent calls Enable Call Waiting
7. Temperature Humidity Compliance HandyTone 286 1xRJ45 10Base T 1 GREEN amp RED color Input 100 240VAC Output 5VDC 1200mA UL certified 65mm W 93mm D 27mm H 0 57 lbs 0 26kg 32 104 F 0 40 C 10 95 non condensing FCC CE C Tick Handy Tone 286 User Manual Grandstream Networks Inc 5 Basic Operations 5 1 Get Familiar with Key Pad and Voice Prompt HandyTone 286 stores a voice prompt menu for quick browsing and simple configuration To enter this voice prompt menu simply press the button on the HandyTone 286 or pick up the phone and dial ee The following table shows how to use the voice prompt menu to configure the device Menu Voice Prompt User s Options 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 or Enter 9 to toggle the selection Static IP Mode If user selects Static IP Mode user need configure all the IP address information Main Menu O 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 1s announced Enter 12 digit new IP address 1f in Static IP Mode 03 Subnet IP address Same as Menu option 02 04 Gateway IP address Same as Menu option 02 05
8. VLAN tag Default setting 1s blank And 802 1p priority value contains the value of the priority value If set to Yes the device will ignore any SIP message that does not come from the IP address Source IP in the IP header that it is registered to Default is No This parameter controls whether the IP phone supports the DNS SRV route function 27 HandyTone 286 User Manual Grandstream Networks Inc User ID is phone number SIP Registration Unregister On Reboot Registration Expiration Early Dial Dial Plan Prefix No Key Entry Timeout Use as Send Key Local SIP port Local RTP port Use Random Port If the HandyTone 286 has an assigned PSTN telephone number then this field will be set to Yes Otherwise set it to No If Yes is set a user phone parameter will be attached to the From header in SIP request This parameter controls whether the IP phone needs to send REGISTER messages to the proxy server The default setting is Yes Default is No If set to yes the device will first send registration request to remove previous bindings This parameter allows the user to specify the time frequency in minutes the phone will refresh its registration with the specified registrar The default interval is 3600 seconds or 1 hour The maximum interval is 45 days This parameter controls whether the phone will attempt to send an early INVITE each time a key is pressed w
9. address or domain name of firmware server IP address or domain name of configuration server Default blank If it is configured HT286 rev 3 0 will request the firmware file with the prefix 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 32 digit in Hexadecimal Representation Useful for ITSP to encrypt firmware End user should keep it blank Default NO End user should use default setting If this parameter is set to Yes except for IVR MENU items 1 to 5 the configuration update via keypad is disabled If set to Yes these four fields SIP User ID Authenticate ID Authenticate Password and Name will be included in Basic Settings configuration page Override the MTU size 32 Handy Tone 286 User Manual Grandstream Networks Inc Volume Handset volume adjustment RX is for receiving volume TX is for Amplification transmission volume Default values are OdB for both parameters 6dB generates the highest volume and 6dB generates the lowest volume Call Progress Tones Using these settings users can configure various call progress tone frequencies and cadences according to their country standard By default they are set to North American standard Frequencies should be configured with known values to avoid uncomfortable high pitch sounds ON is the perio
10. default destination port 5060 is used if no port is specified 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 5 2 3 Call Hold While in conversation pressing the flash button on the attached phone 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 5 2 4 Call Waiting 11 Handy Tone 286 User Manual Grandstream Networks Inc 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 52 5 Call Transfer 5 2 5 1 Blind Transfer Assuming that call party A and B are in conversation A wants to Blind Transfer B to C 1 A presses FLASH on the analog phone or Hook Flash for old model phones to get a dial tone 2 Then A dials 87 then dials C s number and then or wait for 4 seconds 3 A can hang up NOTE Enable Call Feature has to be set to Yes in web configuration page A can hold on to the phone and wait for one of the three following behaviors e A quick confirmation tone temporari
11. embedded Web server that will respond to HTTP GET POST requests It also has embedded HTML pages that allows a user to configure the HandyTone 286 through a Web browser such as Microsoft s IE and AOL s Netscape 6 2 1 Access the Web Configuration Menu First get the IP address of the HandyTone 286 through section 5 1 with menu option 02 Then access the HandyTone 286 s Web Configuration Menu using the following URI http HandyTone IP Address where the HandyTone IP Address is the IP address of the HandyTone 286 NOTE e To type IP address into browser to get into the configuration page please strip out the leading 0 as the browser will parse in octet e g 1f the IP address is 192 168 001 014 please type in 192 168 1 14 16 HandyTone 286 User Manual Grandstream Networks Inc 6 2 2 End User Configuration Once this request 1s entered and sent from a Web browser the IP phone will respond with the following login screen Password HA Login The password is case sensitive with maximum length of 25 characters The factory default password for End User and administrator 1s 123 and admin respectively Only administrator can get access to ADVANCED SETTING configuration page NOTE If you cannot log into the configuration page by using default password please check with the VoIP service provider Most likely the service provider has provisioned the device and configured for you and cha
12. every ordinary phone call or called POT Plain Old Telephone or circuit switched network RTCP Real time Transport Control Protocol defined in REC 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 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 NATS is 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
13. 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 42 Handy Tone 286 User Manual Grandstream Networks Inc 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 human speech is detected from the audio samples VLAN A virtual LAN known as a VLAN is a logically independent netw
14. power adaptor is compliant with UL standard The HandyTone ATA should only operate with the universal power adaptor 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 286 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 Handy Tone 286 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 Powerful digital signal processing DSP to
15. the new code image will then be saved into the Flash If TFTP HTTP fails for any reason e g TFTP HTTP server 1s 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 Firmware upgrade may take as long as 1 to 20 minutes over Internet or just 20 seconds if 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 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 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 TFIP server s default setting from Receive Only to Transmit Only for the firmware upgrade Start the TFTP server in the HandyTone ATA s
16. Anonymous Method Time to ring Special Features Syslog Server Select the Caller ID Scheme to suit the standard of different area e Bellcore North America e CID Canada e DIMF Brazil e DTMF Denmark e DTMF Sweden e ETSI FSK France Germany Norway Taiwan UK CCA e ETSI DTMEF Finland Sweden Select the onhook voltage to suit different area or PBX Select Polarity Reversal to adapt some call charge billing system Default is No This parameter defines the URI or IP address of the NTP server which the IP phone will use to display the current date time If this parameter is set to Yes the device is employing the mechanism to block its ID If it is set to Use from header Callers SIP user ID will be sent as anonymous essentially block the Caller ID from displaying If it is set to User privacy header the SIP INVITE message contains a privacy header and the server blocks the caller ID from the called party Allow user to adjust the ring time of the phone Default is 60 seconds Default 1s Standard Choose the selection to meet some special requirements from Soft Switch vendors like Nortel Broadsoft etc The IP address or URL of System log server This feature 1s especially useful for ITSP Internet Telephone Service Provider 30 HandyTone 286 User Manual Grandstream Networks Inc Syslog Level Session Expiration Min SE Caller Request Timer Callee Request Timer
17. NAT Mapped IP WAN side mapped IP if HandyTone 286 is connected to a NAT router 18 Handy Tone 286 User Manual Grandstream Networks Inc NAT Mapped Port WAN side mapped port if HandyTone 286 is connected to a NAT router Statistical Status Self explainable Please refer to the page displayed e Basic Settings End User A os y not displayed for security protection Password Web Port A defautt for HTTP is 80 IP Address dynamically 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 Preferred DNS server Po fo a o BE ican configured as IP Address 192 168 l 160 Subnet Mask KE EH Po i Default Router Ed ey wi DNS Server 1 rel Ee B l DNS Server 2 El E l Time Zone IA E No Ell ves if set to Yes display time will be 1 hour ahead of _ normal time Daylight Savings Time Optional Rule 19 HandyTone 286 User Manual Grandstream Networks Inc 20 End User Password Web Port IP Address DHCP hostname DHCP domain DHCP vendor class ID PPPoE account ID PPPoE password Time Zone Daylight Time Savings This contains the password to access the Web Configuration Menu This field is case sensitive with max 25 characters This is the device s internal HTTP server port Default is 80 There are 2 modes unde
18. P 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 appropriate responses in the form of voice fax callback e mail and perhaps other media MTU A Maximum
19. TU Size rx Pg defaut py Volume Amplification Frequency 1 Frequency 2 ON x10ms OFF x 1 ms Call Progress Tones Hz Hz C1 C2 C3 C1 C2 C3 Dial Tone EE Ga EE Recall Dial Tone OT ZO OS 25 Handy Tone 286 User Manual Grandstream Networks Inc Admin Password SIP Server Outbound Proxy SIP User ID Authenticate ID Authenticate Password Name Home NPA Preferred Vocoder Message Waiting Confirmation Audible Ringing Busy Tone Reorder Tone Receiver Offhook Tone Administrator password Only administrator can configure the Advanced Settings page Password field is purposely left blank for security reason after clicking update and saved The maximum password length is 25 characters This field contains the URI string or the IP address e g sip my voip provider com 192 168 1 200 5066 This field contains the URI string or the IP of the outbound proxy If there is no outbound proxy this field SHOULD be left blank If it is not blank all outgoing requests will be sent to this outbound proxy This field contains the user part of the SIP address for this phone e g if the SIP address is sip my_user_id my_provider com then the SIP User ID is my_user_id Please do NOT include the preceding sip scheme or the host portion of the SIP address in this field It is given by VoIP service provider SIP service subscribe
20. 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 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 41 HandyTone 286 User Manual Grandstream Networks Inc 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 mainly with cable modem and DSL services PSTN Public Switched Telephone Network 1 e the phone service we use for
21. User Manual 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 if 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 C 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 O o gt The device will reboot automatically and restore to factory default setting 37 Grandstream Networks Inc Handy Tone 286 User Manual Grandstream Networks Inc 9 Glossary of Terms ADSL Asymmetric Digital Subscriber Line Modems attached to twisted pair copper wiring that transmit from 1 5 Mbps to 9 Mbps downstream to the subscriber and from 16 kbps to 800 kbps upstream depending on line dista
22. User Manual Handy lone 286 Rev 3 0 Analog Telephone Adaptor For Firmware Version 1 0 8 32 Grandstream Networks Inc www grandstream com Handy Tone 286 User Manual Grandstream Networks Inc Table of Contents 1 AWNELCOME iai ainia 4 2 INSTALLATION rE EEOAE TANEET TAE TESSE 5 3 WHAT IS INCLUDED IN THE PACKAGE esesescccesesesecececcocccccosececececeoeccceceocococcsesesesesesesescoceces 6 34 SAFE eC OMPLIANCE A OE 6 P WARRANT A A aussie teu E eeadanet 6 4 PRODUCT OVERVIEW serii N a EESE Es 7 A GS a IN 7 A HARDWARE SPECIES db leas 8 S BASICOPERATIOONS casa o iiad 9 5 1 GET FAMILIAR WITH KEY PAD AND VOICE PROMPT sscesceccecceccscesceccecceccscescecceceecescesceceecens 9 SA AE PONE E e IIA e o eE 10 524 Calling Phone or Extension Numbers Sana 10 I FZ DRA A A 1 O CAO 4 NANA NN IA A IN 1 Dele AAA A A A cusccvwcuivntns 11 JAI TALIS 21 2222225074 28 225 02 As sains A A 12 5 2 5 1 E eames ee este a NTIS ee ne BO eS Pen Ye ee SR ee mento oe EME ONE Rarer OO a MUTA TTY SME TEE on 2 Car Re A 12 5 2 5 2 A vary E a ae a 12 IO OCO TCINE is 90009 099777 4000 RO A ARAE 00077 DEMA 2822098576 13 5 2 6 1 ESA E a Ce A Gre Si N AA ASE eA AO An O 22 NOSA NE EE ala ale ce 13 3202A Novebell style CONTENER ES AAA An 13 LE XCALI FEATURES esla ena R a eee eon AT eee 13 ate O EN 14 ID WED LIGHT PATTERN INDICATION odds 14 6 CONFIGURA TION GUIDE siii caian 16 6 1 CONFIGURING HANDYTONE 286 IP THROUGH VOICE PROMPT
23. d 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 vocoder in the above vocoder Preference List or the actual used payload type negotiated between the 2 conversation parties at run time e g if the first vocoder 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 vocoder 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 phone will use and save the maximum allowed value for the corresponding first vocoder choice The maximum value for PCM is 10 x10ms frames for G726 it is 20 x10ms frames for G723 it is 32 x30ms frames for G729 G728 64 x10ms and 64 x2 5ms frames respectively T 38 Auto Detect FoIP by default or Pass Through must use codec PCMU PCMA This field defines the layer 3 QoS parameter which can be the value used for IP Precedence or Diff Serv Default value 1s 48 This setting includes two fields The 802 1Q VLAN Tag contains the value used for layer 2
24. d 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 Up to three cadences are supported 6 2 4 Saving the Configuration Changes Once a change is made the user should press the Update button in the Configuration Menu The IP phone 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 power cycle the HandyTone 286 after seeing the above message 6 2 5 Rebooting the HandyTone 286 from Remote The administrator of the phone can remotely reboot the phone by pressing the Reboot button at the bottom of the configuration menu Once done the following screen will be displayed to indicate that rebooting is underway 33 Handy Tone 286 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 phone after waiting for about 30 seconds 6 3 Configuration through a Central Server Grandstream HandyTone ATA can be automatically configured from a central provisioning system When HandyTone ATA boot up it will send TFTP or HTTP request to download configuration file cfg000b82xxxxxx w
25. ensure superb audio quality 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 G 729A and iLBC Support Caller D name display or block Call waiting caller ID Hold Call Waiting Flash Call Transfer 3 way conference on Rev 2 0 Call Forward in band and out of band DTMF etc Support fax pass through for PCMU and PCMA and T 38 FoIP Fax over IP 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 encrypted configuration files through TFTP or HTTP server Support firmware upgrade via TFTP or HTTP Support SIP Session Timer Support Syslog on Rev 2 0 Ultra compact wallet size and lightweight design great companion for travelers Compact lightweight Universal Power adaptor 4 2 Hardware Specification Grandstream Networks Inc HandyTone 286 User Manual The table below lists the hardware specification of HandyTone 286 Model LAN interface Button LED Universal Power Adaptor Dimension Weight Operating
26. ess 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 FODN 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 FXO Foreign eXchange Office 39 Handy Tone 286 User Manual Grandstream Networks Inc 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 is 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 An FXS device will allow any FXO device to operate as if it were connected to the phone company This makes your PBX
27. for all subsequent calls HandyTone 286 User Manual Grandstream Networks Inc Disable Call Waiting Per Call Enable Call Waiting Per Call 72 Unconditional Call Forward To use this feature dial 72 and get the dial tone Then dial the forward number and for a dial tone then hang up 73 Cancel Unconditional Call Forward To cancel Unconditional Call Forward dial 73 and get the dial tone then hang up 90 Busy Call Forward To use this feature dial 90 and get the dial tone Then dial the forward number and for a dial tone then hang up 9 Cancel Busy Call Forward To cancel Busy Call Forward dial 91 and get the dial tone then hang up 92 Delayed Call Forward To use this feature dial 92 and get the dial tone Then dial the forward number and for a dial tone then hang up 93 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 1f there 1s a call waiting indication When in conversation without an incoming call this action will switch to a new channel for a new call 54 Fax HandyTone 286 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 me
28. hen 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 most likely be rejected by the proxy with a 404 Not Found error Please note that this feature is NOT designed to work with and should NOT be enabled for direct IP to IP calling This value contains the dial plan prefix string typically an ASCII numeric string If it 1s not blank then this string will added to the dialed number Default is 4 seconds 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 is essentially equivalent to the Re Dial key If set to No this key will then be included as part of the dial string to be sent out This parameter defines the local SIP port the IP phone will listen and transmit on The default value is 5060 This parameter defines the local RTP RTCP port pair the IP phone will listen and transmit on It is the base RTP port for channel 0 When configured channel O wil
29. here QOOb82xxxxxx is the MAC address of the HandyTone ATA The configuration file can be loaded into devices via TFTP or HTTP from the central provisioning server so the service provider or an enterprise with large deployment of HandyTone ATAs can easily manage the configuration and service provision to individual devices remotely Grandstream provides a licensed provisioning system called GAPS that can be used to support automated configuration of HandyTone ATA GAPS Grandstream Automated Provisioning System uses enhanced NAT friendly TFTP or HTTP thus no NAT issues and other communication protocols to communicate with each individual HandyTone ATA for firmware upgrade remote reboot etc Grandstream provide GAPS Grandstream Automated Provisioning System service to VoIP service providers It could be either simple redirection or with certain special provisioning settings Initially upon booting up Grandstream devices by default point to Grandstream provisioning server GAPS based on the unique MAC address of each device GAPS provision the devices with redirection settings so that they will be redirected to customer s TFTP or http 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
30. if you were calling from a regular analog phone 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 1f e Both HandyTone ATA and other VoIP Device e another HandyTone ATA or Budgetone SIP phone or other VoIP unit have public IP addresses or e Both HandyTone ATA and other VoIP Device are on the same LAN using private IP addresses or e Both HandyTone ATA and other VoIP Device can be connected through a router using public or private IP addresses with necessary port forwarding or DMZ To make a direct IP call first pick up the analog phone or turn 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 0 160 the dialing convention is Voice Prompt with option 47 then 192168000160 followed by pressing the key if it is configured as a send key or wait 4 seconds In this case the
31. ion and dial tone plays in this state lt 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 Once all digits are accumulated the device will automatically process them e For IP address input omit the dot and enter the digits directly add O for those octets with less than three digits e g IP 192 168 1 10 key in 192168001010 e Key entry cannot be deleted but the phone may prompt error once it is detected 5 2 Make Phone Calls 5 2 1 Calling Phone or Extension Numbers To make a phone or extension number call a Dial the number directly and wait for 4 seconds Default No Key Entry Timeout Or b Dial the number 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 10 Handy Tone 286 User Manual Grandstream Networks Inc you phone is assigned with a PSTN like number such as 6265556789 most likely you just follow the rule to dial 16266667890 as
32. l 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 is 5004 This parameter when set to Yes will force random generation of both the local SIP and RTP ports This is usually necessary when multiple IP phones are behind the same NAT 28 HandyTone 286 User Manual Grandstream Networks Inc NAT Traversal keep alive interval Use NAT IP Proxy Require SUBSCRIBE for MWI Offhook Auto Dial Enable Call Feature Use Bell style 3 way conference Disable Call Waiting Send DTMF DTMF Payload Type Send Flash Event FXS Impedance This parameter defines whether the phone NAT traversal mechanism will be activated or not If activated by choosing Yes and a STUN server is also specified then the phone will behave according to the STUN client specification Under this mode the embedded STUN client inside the phone will attempt to detect 1f and what type of firewall NAT it is behind by sending appropriate request to the specified STUN server If this field is set to Yes with no specified STUN server then the phone will only periodically every 20 seconds by default send a blank UDP packet with no payload data to the SIP server to keep the mapped port open on the NAT The HandyTone 286 sends a UDP package to the SIP server periodically in order to keep the port open on the router This parameter defines the inte
33. ly using the call waiting indication tone followed by a dial tone This indicates the transfer 1s successful transferee has received a 200 OK from transfer target At this point A can either hang up or make another call e 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 e Busy tone keeps playing This means we have failed to receive the second 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 1s 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 Assuming 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 or Hook Flash for old model phones to get a dial tone 2 A then dial C s number then 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 12 Handy Tone 286 User Manual Grandstream Networks
34. munications A standard developed by the European Telecommunication Standard Institute from 1988 governing pan European digital mobile telephony DECT covers wireless PBXs telepoint residential cordless telephones wireless access to the public switched telephone network Closed User Groups CUGs Local Area 38 Handy Tone 286 User Manual Grandstream Networks Inc 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 extension without going through an attendant or auto attendant DSP Digital Signal Processing Using computers to proc
35. nce 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 IPv4 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 or compressed Lossy compression algorithms ordinarily decimate while subsampling DECT Digital Enhanced Cordless Telecom
36. nged the default password e Status Page MAC Address 00 0B 82 08 74 D4 IP Address 10 10 13 195 Product Model HT286 REV 3 0 Program 1 0 8 19 Bootloader 1 0 8 11 HTML 1 0 8 19 VOC Software Version 919 System Up Time 0 day s O hour s O minute s Registered Yes PPPoE Link Up disabled NAT detected NAT type is full cone NAT Mapped IP 123 45 6 789 Handy Tone 286 User Manual Grandstream Networks Inc NAT Mapped Port Total Inbound Calls Total Outbound Calls Total Missed Calls Total Call Time in minutes Total SIP Message Sent Total SIP Message Received Total RTP Packet Sent Total RTP Packet Received Total RTP Packet Loss MAC Address The device ID in HEX format This is very important ID for ISP troubleshooting IP Address IP address assigned on the device Product Model This field contains the product model info Software Version Program This should be consistent with the firmware version Bootloader version number for bootloader it could be lower than program version HTML This should be consistent with the firmware version VOC This is the codec program normally not changed System Uptime This shows system up time since last reboot Registered This shows whether the unit is registered to voip service provider s server PPPoE Link Up This shows whether the PPPoE is up if connected to DSL modem NAT This shows what kind of NAT the HandyTone 286 is behind
37. o 10 20 32 64 for G711 G726 G723 other codecs respectively Eras Auto Detect E pass T hrough Ea Diff Serv or Precedence value 802 1Q VLAN Tag 802 1p priority value 207 Eno 4 No G No Yes ves Elo Edy Elo 3600 in seconds default 1 hour max 45 days L Yes L Yes E No E use Yes only 1f proxy supports 484 response E No E this prefix string is added to each dialed number a Yes O n seconds default 1s 4 seconds Elo Eyes if set to Yes will function as the Dial key 5060 default 5060 5004 1024 65535 default 5004 E No No ves STUN server is p URI or IP port 20 in seconds default 20 seconds ee lt c in SIP SDP message if specified 23 a Yes Handy Tone 286 User Manual Grandstream Networks Inc Proxy Require A SUBSCRIBE for MWI G No do not send SUBSCRIBE for Message Waiting Indication Yes send Yes send periodical SUBSCRIBE for Message Waiting Indication Offhook Auto Dial A se ID extension to dial automatically when O Enable Call Features Bi vos if Yes Call Forwarding amp Call Waiting Disable are supported locally Use Bell style 3 way Conference Disable Call Waiting Send DTMF DTMF Payload Type Send Flash Event Onhook Threshold FXS Impedance Caller ID Scheme Onhook Voltage No Y Yes reverse polarity upon call establishment and
38. omatic 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 hour 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 the iteration of the weekday 1st Sunday 3rd Tuesday etc If weekday is 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 Handy Tone 286 User Manual Grandstream Networks Inc 6 2 3 Advanced User Configuration To login to the Advanced User Configuration page follow the instruction in section 6 2 1 to get to the following login page The password is ca
39. ork 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 43
40. r s Authenticate ID used for authentication It can be identical to or different from SIP User ID and given by VolP service provider SIP service subscriber s account password It is given by VoIP service provider SIP service subscriber s name which will be used for Caller ID display Local area code for North American Dial Plan HandyTone 286 supports up to 7 different vocoder types including G711 ulaw PCMU G711 alaw PCMA G723 G729A G726 32 and iLBC Depending on the product model some of these vocoders may not be provided in standard release A user can configure vocoders in a preference order that will be offered in SIP INVITE message 26 HandyTone 286 User Manual Grandstream Networks Inc G723 Rate iLBC frame size iLBC payload type Silence Suppression Voice Frames per TX Fax Mode Layer 3 QoS Layer 2 QoS Allow incoming SIP messages from SIP proxy only Use DNS SRV This defines the encoding rate for G723 vocoder By default 6 3kbps rate is chosen This defines the size of the LBC codec frame The default setting is 20ms This defines the iLBC payload type The default setting is 97 The valid range is between 96 and 127 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 disable
41. r which the IP phone can operate If DHCP mode is enabled then all the field for the Static IP mode are not used even though they are still saved in the Flash memory and the IP phone will acquire its IP address from the DHCP server in the network To use PPPoE feature please set the PPPoE account settings if the HT 286 is connected directly to a DSL modem The HT 286 will attempt to establish a PPPoE session if any of the PPPoE fields is set If Static IP mode is selected then the IP address Subnet Mask Default Router IP address DNS Server mandatory DNS Server 2 optional fields will need to be configured This option specifies the name of the client This field 1s 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 PPPoE username Fill this field if your ISP requires you to use a PPPoE Point to Point Protocol over Ethernet connection PPPoE account password This parameter controls how date time is displayed according to the specified time zone This parameter controls whether the displayed time will be daylight savings 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 Aut
42. refix and Postfix Starting from firmware version 1 0 7 11 for HandyTone 286 Rev 3 0 adding prefix and postfix for both firmware and configuration file is supported 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 1s set to Yes the device will only issue firmware upgrade request 1f there are changes in the firmware Prefix or Postf1x 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 1f 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 36 HandyTone 286
43. rval time that HT286 send the UDP package The default setting is 20 second NAT IP address is used in SIP SDP message Default is blank SIP Extension to notify SIP server that the unit is behind the NAT Firewall Default is No When set to Yes a SUBSCRIBE for Message Waiting Indication will be sent periodically This parameter allows the 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 phone will automatically append the and the host portion of the corresponding SIP address Default is YES If set to Yes call features are supported locally such as call waiting transfer 3 way conference etc Conference mode default option is No If set to yes the feature code for coference 23 would be disabled Default is No This parameter specifies the mechanism to transmit DTMF digit There are 3 modes supported in audio which means DTMF is combined in audio signal not very reliable with low bit rate codec via RTP RFC2833 or via SIP INFO This parameter sets the payload type for DTMF using RFC2833 Default is NO If set to yes flash will be sent as DTMF event Selects the impedance of the analog telephone connected to the Phone port 29 Handy Tone 286 User Manual Grandstream Networks Inc Caller ID Scheme Onhook Voltage Polarity Reversal NTP server Send Anonymous
44. s can be downloaded from http www grandstream com DOWNLOAD Configuration_Tool For details on how GAPS works please refer to the documentation of GAPS product 34 Handy Tone 286 User Manual Grandstream Networks Inc 7 Software Upgrade with TFTP 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 TF TP 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 8 19 e g 168 75 215 189 NOTES 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 Once a Firmware Server Path 1s 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 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
45. se sensitive with a maximum length of 25 characters and the factory default password for Advanced User is admin O e not displayed for security protection SIP Server sip mycompany com gt gt gt q g Sip mycompany com or IP address A gt proxy myprovider com or IP address if any SIP User ID AAA user part of an SIP address Authenticate ID 123456789 gt gt gt cam be identical to or different from SIP User ID A os y not displayed for security protection Name A o e g John Doe Admin Password Outbound Proxy Authenticate Password Advanced Options Preferred Vocoder in listed order choice 2 choice 1 choice 3 choice 4 choice 5 choice 6 choice 7 G 23 rate Edo 3kbps encoding rate Eds kbps encoding rate 22 Handy Tone 286 User Manual Grandstream Networks Inc iLBC frame size LBC payload type Silence Suppression Voice Frames per TX Fax Mode Layer 3 QoS Layer 2 QoS Allow incoming SIP messages from SIP proxy only Use DNS SRV User ID is phone number SIP Registration Unregister On Reboot Register Expiration Early Dial Allow outgoing call without Registration Dial Plan Prefix No Key Entry Timeout Use as Dial Key local SIP port local RTP port Use random port NAT Traversal keep alive interval Use NAT IP Eons o 30ms between 96 and 127 default 1s 97 E No E ves cm up t
46. te party supports session timer If selecting Yes the phone will use session timer when it receives inbound calls with session timer request 31 HandyTone 286 User Manual Grandstream Networks Inc Force Timer UAC Specify Refresher UAS Specify Refresher Force INVITE Firmware Upgrade and Provisioning Firmware Server Path Config Server Path Firmware File Prefix Firmware File Postfix Config File Prefix Config File Postfix Automatic Upgrade Firmware Key Authenticate Conf File Lock keypad update Allow conf SIP Account in Basic Settings Override MTU size 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 HTTP Firmware upgrading may take up to 10 minutes depends on network environment Do not interrupt the firmware upgrading process IP
47. 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 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 Protocol A packet based protocol for delivering data across networks 40 Handy Tone 286 User Manual Grandstream Networks Inc IP PBX IP based Private Branch Exchange I
48. thod by selecting Fax mode to be T 38 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 will need to select all the Preferred Codecs to be PEMU PCMA 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 HandyTone 286 fails to register Button flashes every 2 seconds if SIP server is HandyTone 286 User Manual Grandstream Networks Inc configured Firmware Upgrading Button flashes every 2 seconds Device Malfunctions Red light steady on GREEN LED mostly indicates normal working status Button flashes every 2 seconds Green light steady on 15 Handy Tone 286 User Manual Grandstream Networks Inc 6 Configuration Guide 6 1 Configuring HandyTone 286 IP through Voice Prompt 6 1 1 DHCP Mode Follow section 5 1 with voice menu option 01 to enable HandyTone 286 to use DHCP 6 1 2 STATIC IP Mode Follow section 5 1 with voice menu option 01 to enable HandyTone 286 to use STATIC IP mode then use option 02 03 04 05 to set up IP address Subnet Mask Gateway and DNS server respectively 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 286 with Web Browser HandyTone 286 has an
49. 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 35 Handy Tone 286 User Manual Grandstream Networks Inc the TFTP server is connected to the device s LAN port the firmware upgrade will not work by design 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 is 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 up or reboots 1t will issue request for configuration file named C SXXXXXXXXXXXX Where XXXXXXXXXXxXx is the MAC address of the device 1 e cf2 000b820102ab The configuration file name should be in lower cases 73 Firmware and Configuration File P

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