Grandstream Networks HANDYTONE HandyTone-286 User's Manual
Contents
1. 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 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 29 HandyTone 286 User Manual Grandstream Networks Inc 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 The device is rebooting now You may relogin by clicking on the link below in 30 seconds Click to relogin 30 HandyTone 286 User Manual Grandstream Networks Inc At this point the user can relogin to the phone after waiting for about 30 seconds 6 3 C2. 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 Built in router NAT Gateway and DMZ port forwarding Can also be configured to function as a two Ethernet ports bridge NAT function is disabled Powerful digital signal processing DSP to 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 as well as G 729A and iLBC Support Caller ID 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
3. 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 Firmware upgrade may take as long as 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 se
4. 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 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 HandyTone 286 User Manual Grandstream Networks Inc 4 1 4 Product Overview
5. 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 templates 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 31 HandyTone 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 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 8 16 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 is set user needs to update the settings and reboot the device If the configured firmware server is found and a new code image is available
6. call establishment and termination 21 HandyTone 286 User Manual Grandstream Networks Inc NTP Server tmenistov URI or IP address Send Anonymous 9 No G Yes caller ID will be blocked if set to Yes Anonymous Method Use From Header G Use Privacy Header Special Feature Standard Syslog Server Syslog Level NONE im Session Expiration in seconds default 180 seconds Min SE in seconds default and minimum 90 seconds Caller Request Timer Yes 9 No Request for timer when making outbound calls Callee Request Timer Yes No When caller supports timer but did not request one Force Timer Yes No Use timer even when remote party does not support UAS Specify Refresher UAC UAS When UAC did not specify refresher tag G G G UAC Specify Refresher EX Yac Uas Omit Recommended E G Force INVITE Yes No Always refresh with INVITE instead of UPDATE Firmware Upgrade and ieee Upgrade Via G TFTP o HTTP Provisioning fm grandstream convgs Firmware Server Path A fm grandstream convgs Config Server Path J Firmware File Prefix Firmware File Postfix Config File Prefix Config File Postfix Automatic Upgrade o No G Yes check for upgrade every 008 minutes default 7 days o Always Check for New Firmware G Check New Firmware only when F W pre suffix changes Firmware Key in Hexadecimal Representation Authenticate Conf File ce No G Yes cfg file wou
7. 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 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 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 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 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 mod
8. 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 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 This defines the encoding rate for G723 vocoder By default 6 3kbps rate is chosen This defines the size of the iLBC codec frame The default setting is 20ms 23 HandyTone 286 User Manual Grandstream Networks Inc 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 User ID is phone number 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 disabled 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 t
9. 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 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 24 HandyTone 286 User Manual Grandstream Networks Inc 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 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 when a user dials a number If set to Yes an INVITE is sent using the dial number collected thus far Otherwise no INVITE is sent until the Re Dial button is pressed or after a
10. 92 and get the dial tone Then dial the forward number and for a dial tone then hang up 793 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 there is a call waiting indication When in conversation without an incoming call this action will switch to a new channel for a new call 5 4 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 method 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 PCMU PCMA 13 HandyTone 286 User Manual Grandstream Networks Inc 5 5 LED Light Pattern Indication Following are the LED light pattern indications RED LED always indicates not abnormal status Button flashes every 2 seconds if DHCP is configured Button flashes every 2 seconds if SIP server is configured Firmware Upgrading Button flashes every 2 seconds Device Malfunctions Red light steady on 14 HandyTone 286 User Manual Grandstream Networks Inc 6 Configuration Guide 6 1 Configuring Han
11. CTORY DEFAULT SETTING 34 HandyTone 286 User Manual Grandstream Networks Inc 9 GLOSSARY OF TERMS 35 HandyTone 286 User Manual Grandstream Networks Inc 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 pdf HandyTone 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 RJ11 Telephone 10M Ethernet 5V 1200mA Interconnection Diagram of the HandyTone 286 Analog Phone Internet ADSL Cable Modem Ethernet PHONE Cordless Phone J HandyTone 286 User Manual Grandstream Networks Inc 3 What is Included in the Package The HandyTone 286 package contains 1 One HandyTone
12. IP phone will respond with the following login screen Password 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 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 changed 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 16 Bootloader 1 0 8 9 HTML 1 0 8 16 VOC Software Version 0010 System Up Time 0 day s 0 hour s 0 minute s Registered Yes PPPoE Link Up disabled NAT detected NAT type is full cone NAT Mapped IP 67 153 142 80 HandyTone 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 Versi
13. 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 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 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
14. User Manual Handy Tone 286 Rev 3 0 Analog Telephone Adaptor For Firmware Version 1 0 8 16 Grandstream Networks Inc WWW 2 randstream com CE stun HandyTone 286 User Manual Grandstream Networks Inc Table of Contents 1 WELCOME 4 2 INSTALLATION 5 3 WHAT IS INCLUDED IN THE PACKAGE 3 1 SAFETY COMPLIANCES 3 2 WARRANTY 6 6 6 4 PRODUCT OVERVIEW 7 7 8 9 4 1 KEY FEATURES 4 2 HARDWARE SPECIFICATION 5 BASIC OPERATIONS 5 1 GET FAMILIAR WITH KEY PAD AND VOICE PROMPT 9 5 2 MAKE PHONE CALLS 10 5 2 1 Calling Phone or Extension Numbers 10 5 2 2 Direct IP Calls 10 5 2 3 Call Hold 11 5 2 4 Call Waiting Il 5 2 5 Call Transfer 11 5 2 6 3 way Conferencing 12 5 3 CALL FEATURES 12 5 4 FAX 13 5 5 LED LIGHT PATTERN INDICATION 14 6 CONFIGURATION GUIDE 15 6 1 CONFIGURING HANDYTONE 286 IP THROUGH VOICE PROMPT 15 6 1 1 DHCP Mode 15 6 1 2 STATIC IP Mode 15 6 1 3 TFTP Server Address 15 6 2 CONFIGURING HANDYTONE 286 WITH WEB BROWSER 15 6 2 1 Access the Web Configuration Menu 15 6 2 2 End User Configuration 16 6 2 3 Advanced User Configuration 19 6 2 4 Saving the Configuration Changes 30 6 2 5 Rebooting the HandyTone 286 from Remote 30 6 3 CONFIGURATION THROUGH A CENTRAL SERVER 31 7 SOFTWARE UPGRADE WITH TFTP 32 7 1 FIRMWARE UPGRADE THROUGH TFTP HTTP 32 7 2 CONFIGURATION FILE DOWNLOAD 33 7 3 FIRMWARE AND CONFIGURATION FILE PREFIX AND POSTFIX 33 7 4 MANAGING FIRMWARE AND CONFIGURATION FILE DOWNLOAD 33 8 RESTORE FA
15. Yes G Yes G No Yes o No i in seconds default 1 hour max 45 days 20 HandyTone 286 User Manual Grandstream Networks Inc Early Dial E No G Yes use Yes only if proxy supports 484 response Dial Plan Prefix this prefix string is added to each dialed number No Key Entry Timeout 4 in seconds default is 4 seconds Use as Dial Key G No Q Yes if set to Yes will function as the Dial key local SIP port 5060 default 5060 local RTP port 5 4 1924 65535 default 5004 Use random port W No G Yes NAT Traversal C No 9 Yes STUN server is URI or IP port keep alive interval au in seconds default 20 seconds Use NAT IP used in SIP SDP message if specified Proxy Require SUBSCRIBE yon MVE jE No do not send SUBSCRIBE for Message Waiting Indication G Yes send periodical SUBSCRIBE for Message Waiting Indication Offhook Auto Dial User ID extension to dial automatically when offhook Bete Cat TE ar ii E No o Yes if Yes Call Forwarding amp Call Waiting Disable are supported locally Disable Call Waiting No G Wes Send DTMF in audio via RTP RFC2833 via SIP INFO DTMF Payload Type Send Flash Event E No G Yes Flash will be sent as a DTMF event if set to Yes Onhook Threshold 800 ms z FXS Impedance 600 Ohm North America Caller ID Scheme Bellcore North America Onhook Voltage 36V Polarity Reversal o No G Yes reverse polarity upon
16. and lightweight design great companion for travelers Compact lightweight Universal Power adaptor HandyTone 286 User Manual Grandstream Networks Inc 4 2 Hardware Specification The table below lists the hardware specification of HandyTone 286 Model HandyTone 286 LAN interface 1xRJ45 10Base T Button 1 LED GREEN amp RED color Universal Input 100 240VAC Power Adaptor Output 5VDC 1200mA UL certified Dimension 65mm W 93mm D 27mm H Weight 0 57 lbs 0 26kg Operating 32 104 F Temperature 0 40 C Humidity 10 95 non condensing Compliance FCC CE C Tick HandyTone 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 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 Ol Static IP Mode or Enter 9 to toggle the selection Dynamic IP Mode If user selects Static IP Mode user need configure the all IP address information through menu 02 to O05 If user selects Dynamic IP Mode the device will retrie
17. arable 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 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 38 HandyTone 286 User Manual Grandstream Networks Inc OBP SBCs to allow the use of VoIP protocols from private networks w
18. bout 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 is 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 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 is 5004 This parameter when set
19. cheduled 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 33 HandyTone 286 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 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 ao Pf The device will reboot automatically and restore to factory default setting 34 HandyTone 286 User Manual Grandstream
20. d Wide Web protocol that performs the request and retrieve functions of a server Internet Protocol A packet based protocol for delivering data across networks 37 HandyTone 286 User Manual Grandstream Networks Inc 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 comp
21. dyTone 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 TETP 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 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 if the IP address is 192 168 001 014 please type in 192 168 1 14 15 HandyTone 286 User Manual Grandstream Networks Inc 6 2 2 End User Configuration Once this request is entered and sent from a Web browser the
22. e 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 address or domain name of firmware server IP address or domain name of configuration server Default blank If it is configured HT486 rev 2 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
23. efault 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 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 26 HandyTone 286 User Manual Grandstream Networks Inc Caller ID Scheme Onhook Voltage Polarity Reversal NTP server Send Anonymous Anonymous Method Special Features Syslog Server Select the Caller ID Scheme to suit the standard of different area e Bellcore North America e CID Canada e DTMF Brazil e DTMF Denmark e DTMF Sweden e ETSI FSK France Germany Norway Taiwan UK CCA e ETSI DTMF Finland Sweden Select the onhook
24. el phones to get a dial 2 ak 23 followed by C s number then or wait for 4 seconds 3 If C answers the call then A presses flash to bring B C in the conference 4 If C does not answer the call A can press flash back to talk to B 5 3 Call Features 12 HandyTone 286 User Manual Grandstream Networks Inc Following table shows the call features of HandyTone 286 Key Call Features 23 3 way conference code 87 Bland transfer 30 Block Caller ID for all subsequent calls 31 Send Caller ID for all subsequent calls 67 Block Caller ID per call 82 Send Caller ID per call 50 Disable Call Waiting for all subsequent calls S1 Enable Call Waiting for all subsequent calls 70 Disable Call Waiting Per Call 71 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 Q 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
25. ess 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 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 39 HandyTone 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 fo
26. essing 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 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 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 2 E 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 11 HandyTone 286 User Manual Grandstream Networks Inc e 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 e A quick busy tone followed by a restored call on supported platforms only This means the transferee has received a 4xx response
27. evel 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 remote party supports session timer If selecting Yes the phone will use session timer when it receives inbound calls with session timer request 28 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 Firmwar
28. firm an option e All entered digit sequences have known lengths 2 digits for menu option and 12 digits for IP address Once all digits are accumulated it automatically processes them 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 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 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 e Both HandyTone ATA and other VoIP Device i e another Hand
29. for Advanced User is admin Admin Password purposely not displayed for security protection SIP Server aud be PAM e g Sip mycompany com or IP address Outbound Proxy R e g proxy myprovider com or IP address if SIP User ID 12945678 the user part of an SIP address HandyTone 286 User Manual Grandstream Networks Inc Authenticate ID Authenticate Password Name Advanced Options Preferred Vocoder in listed order 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 User ID is phone number SIP Registration Unregister On Reboot Register Expiration ee can be identical to or different from SIP User ID purposely not displayed for security protection salute optional e g John Doe current setting is POMU current setting is POMA current setting is G723 r current setting is iLBC current setting is POMU o 6 3kbps encoding rate choice 1 choice 2 choice 3 choice 4 choice 5 choice 6 choice 7 G 5 3kbps encoding rate o 20ms G 30ms o er 96 and 127 default is 97 9 No G Yes g up to 10 20 32 64 for G711 G726 G723 other codecs respectively 9 T 38 Auto Detect G Pass Through Diff Serv or Precedence value 802 1Q VLAN Tag 802 1p priority value 0 7 G Yes G
30. ime 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 is 48 This setting includes two fields The 802 1Q VLAN Tag contains the value used for layer 2 VLAN tag Default setting is blank And 802 1p priority value contains the value of the priority value If set to Yes the device will ignore any SIP message
31. it will issue request for configuration file named C EXXXXXXXXXXXX Where XXXXXXXXXxxx is the MAC address of the device i e cfg000b820102ab The configuration file name should be in lower cases 7 3 Firmware and Configuration File Prefix and Postfix Starting from firmware version 1 0 7 11 for HandyTone 486 Rev 2 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 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 s
32. ith 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 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 designed for voice transmission and uses fewer resources and is considerably l
33. ld be authenticated before acceptance if set to Yes 22 HandyTone 286 User Manual Grandstream Networks Inc Lock keypad update o No G Yes configuration update via keypad is disabled if set to Yes Allow conf SIP Account E C in Basic Settings No Yes Override MTU Size 0 Admin Password SIP Server Outbound Proxy SIP User ID Authenticate ID Authenticate Password Name Preferred Vocoder G723 Rate iLBC frame size 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 subscriber s Authenticate ID used for authentication It can be identical to or different from SIP User ID and given by VoIP
34. o FXS and the PSTN 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 the POTS PSTN for the phone The FXS Interface connects to FXO devices by an FXO interface of course 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 H 323 HTTP IP 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 A suite of standards for multimedia conferences on traditional packet switched networks Hyper Text Transfer Protocol the Worl
35. on 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 NAT Mapped IP WAN side mapped IP if HandyTone 286 is connected to a NAT router 17 HandyTone 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 o s Password purposely not displayed for security protection IP Address E dynamically assigned via DHCP default or PPPoE will attempt PPPoE if DHCP fails and following is non blank PPPoE account ID PPPoE password Preferred DNS PETA i i G statically configured as IP Address KA 168 Jo 160 Subnet Mask e fe 1 Jo Default Router fo fe Je Je DNS Server 1 i 9 Je fo DNS Server 2 4 Jo Jo Time Zone GMT 5 00 US Eastern Time New York Daylight Savings No G Yes if set to Yes display time will be 1 ho
36. onfiguration 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 where OOOb82xxxxxx 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
37. r 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 network Several VLANs 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 40
38. rver 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 32 HandyTone 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 i e P2 is associated with Admin Password in the ADVANCED SETTINGS page Fora detailed parameter list please refer to the corresponding firmware release configuration template When Grandstream Device boots up or reboots
39. s 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 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 FXO Foreign eXchange Office 36 HandyTone 286 User Manual Grandstream Networks Inc FXS DHCP 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 t
40. 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 Telecommunications 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 35 HandyTone 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 translate
41. 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 25 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 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 if 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 interval 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 D
42. ur ahead of Time normal time 18 HandyTone 286 User Manual Grandstream Networks Inc End User This contains the password to access the Web Configuration Menu This field is Password case sensitive with max 25 characters IP Address There are 2 modes under 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 Time Zone This parameter controls how date time is displayed according to the specified time zone Daylight Savings This parameter controls whether the displayed time will be daylight savings Time time or not If set to Yes then the displayed time will be 1 hour ahead of normal time 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 case sensitive with a maximum length of 25 characters and the factory default password
43. ve all IP address information from DHCP server automatically when user reboots the device 02 IP Address IP address The current WAN IP address is announced Enter 12 digit new IP address if in Static IP Mode 03 Subnet IP address Same as Menu option 02 04 Gateway IP address Same as Menu option 02 05 DNS Server IP address Same as Menu option 02 06 TFTP Server IP address Same as Menu option 02 47 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 86 No Voice Messages or If there are voice messages user can dial 9 Voice Messages Pending and dial pre configured phone number to retrieve voice message 99 RESET Enter 9 to reboot the device or Enter MAC address to restore factory default setting For details see section 8 Invalid Entry Automatically return to Main Menu NOTES HandyTone 286 User Manual Grandstream Networks Inc 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 option and dial tone plays in this state e shifts down to the next menu option e returns to the main menu e 9 functions as the ENTER key in many cases to con
44. 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 Default is 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 is especially useful for ITSP Internet Telephone Service Provider 27 HandyTone 286 User Manual Grandstream Networks Inc Syslog Level Session Expiration Min SE Caller Request Timer Callee Request Timer 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 l
45. yTone 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 10 HandyTone 286 User Manual Grandstream Networks Inc 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 6699 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 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 pr
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