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RJ11 Telephone (Phone) RJ45 10M Ethernet LAN - WAN

+5V/1200mA

BUTTON RED LED GREEN LED

RJ11 Phone Line (Line)

Interconnection Diagram of the HandyTone-486:
Internet ADSL/Cable Modem Ethernet Analog Phone

Cordless Phone

Following are the steps to install a HandyTone-486: 1. Connect a standard touch-tone analog telephone (or fax machine) to PHONE port. 2. Connect a PSTN telephone line to LINE port (optional). 3. Insert the Ethernet cable into the WAN port of HandyTone-486 and connect the other end of the Ethernet cable to an uplink port (a router or a modem, etc.) 4. Connect a PC to the LAN port of HandyTone-486. 5. Insert the power adapter into the HandyTone-486 and connect it to a wall outlet. Please follow the instructions in section 6.2.1 to configure the HandyTone-486.
What is Included in the Package
The HandyTone-486 package contains: 1) One HandyTone-486 2) One universal power adaptor 3) One Ethernet cable

Safety Compliances

The HandyTone-486 is compliant with various safety standards including FCC/CE and C-tick. Its power adaptor is compliant with UL standard. The HandyTone-486 should only operate with the universal power adaptor provided in the package.

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-486 and will void the manufacturer warranty.
Caution: Changes or modifications to this product not expressly approved by Grandstream, or operation of this product in any way other than as detailed by this User Manual, could void your manufacturer warranty.
Information in this document is subject to change without notice. No part of this document may be reproduced or transmitted in any form or by any means, electronic or mechanical, for any purpose without the express written permission of Grandstream Networks, Inc.

Product Overview

Make Phone Calls

Calling phone or extension numbers
There are currently two methods to make an extension number call: a) Dial the numbers directly and wait for 4 seconds (Default No Key Entry Timeout). Or b) Dial the numbers directly, and press # (assuming that use # as dial key is selected in web configuration). Examples: To dial another extension on the same proxy, such as 1008, simply pick up 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 the # or wait for 4 seconds.

Direct IP calls

Direct IP calling allows two parties, that is, a HandyTone with an analog phone and another VoIP Device, to talk to each other in an ad hoc fashion without a SIP proxy. This kind of VoIP calls can be made between two parties if: Both HandyTone ATA and other VoIP Device(i.e., another HandyTone ATA or Budgetone SIP phone or other VoIP unit) have public IP addresses, or Both HandyTone ATA and other VoIP Device are on the same LAN using private IP addresses, or 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 to 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 using *4 (encoding for :) followed by the port number. Examples: If the target IP address is 192.168.0.10, the dialing convention is Voice Prompt with option 47, then followed by pressing the # key if it is configured as a send key or wait for more than 5 seconds. If the target IP address/port is 192.168.1.20:5062, then the dialing convention would be: Voice Prompt with option 47, then 192168001020*45062 followed by pressing the # key if it is configured as a send key or wait for 4 seconds. 12

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 Hold party and the bi-directional media will resume.

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.

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. A dials *87 followed by Cs number, 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: A quick confirmation tone (temporarily using the call waiting indication tone) followed by a dial tone. This indicates the transfer is successful (transferee has received a 200 OK from transfer target). At this point, A can either hang up or make another call. A quick busy tone followed by a restored call (on supported platforms only). This means the transferee has received a 4xx response for the INVITE and we will try to recover the call. The busy tone is just to indicate to the transferor that the transfer has failed. Busy tone keeps playing. This means we have failed to receive the 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 13
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 Cs number followed by # (or wait for 4 seconds). 3. If C answers the call, A and C are in conversation. Then A can hang up to complete transfer. 4. If C does not answer the call, A can press FLASH back to talk to B. NOTE: When Attended Transfer failed and A hang up, HandyTone- 486 will ring user A back again to remind A that B is still on the call. A can pick up the phone to restore conversation with B.

3-way Conferencing

HandyTone-496 supports 3-way conference in two styles: star code style or Bellcore style. 5.2.6.1 Star Code Style 3-way Conference Assuming that call party A and B are in conversation. A wants to bring C in a conference: 1. 2. 3. 4. A presses FLASH (on the analog phone, or Hook Flash for old model phones) to get a dial tone. A dials *23 then Cs number then # (or wait for 4 seconds). If C answers the call, then A press flash to bring B, C in the conference. If C does not answer the call, A can press flash back to talk to B.

5.2.6.2 Bellcore Style 3-way Conference Bellcore style 3-way conference is also supported. To do this, user needs to enable Use Bell-style 3way Conference in FXS1 or FXS2 web configuration. Assuming that call party A and B are in conversation. A wants to bring C in a conference: 1. 2. 3. 4. A presses FLASH (on the analog phone, or Hook Flash for old model phones) to get a dial tone. A dials Cs number then # (or wait for 4 seconds). If C answers the call, then A press flash to bring B, C in the conference. If C does not answer the call, A can press flash back to talk to B.
PSTN Pass Through/Life line
HandyTone-486 Rev. 2.0 supports PSTN pass through. User can send and receive PSTN calls with the attached analog phone. To receive PSTN calls, simply make phone off hook when the analog phone rings. 14
To make a PSTN call, press the PSTN access code (*00 by default, or any number configured in web configuration page) to switch to PSTN line and get a dial tone, then dial the PSTN number. When HandyTone-486 is out of power, it will function as a jack. User can use the same analog phone for PSTN calls.

Call Features

Call Features Table (star code)
Following table shows the call features (* code) of HandyTone-486. Key *23 *87 *30 *31 *67 *82 *50 *51 *70 *71 *72 Call Features 3-way conference Blind Transfer Block CallerID (for all-config change) Send CallerID (for all-config change) Block CallerID (per call) Send CallerID (per call) Disable Call Waiting (for all-config change) Enable Call Waiting (for all-config change) Disable Call Waiting. (Per Call) Enable Call Waiting (Per Call) Unconditional Call Forward. To use this feature, dial *72, wait for the dial tone. Then dial the forward number ended with #, wait for dial tone, hang up. Cancel Unconditional Call Forward To cancel Unconditional Call Forward, dial *73 and get the dial tone, then hang up. Busy Call Forward To use this feature, dial *90, wait for the dial tone. Then dial the forward number ended with #, wait for dial tone, hang up. Cancel Busy Call Forward To cancel Busy Call Forward, dial *91 and get the dial tone, then hang up Delayed Call Forward To use this feature, dial *92, wait for the dial tone. Then dial the forward number ended with #, wait for dial tone, hang up. Cancel Delayed Call Forward To cancel this Forward, dial *93 and get the dial tone, then hang up When in conversation, this action will switch to the new incoming call if user heard the call waiting sound. When in conversation and no incoming call heard, this action will switch to a new channel for a new call. 15

Flash/Hook

HandyTone-486 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.
LED Light Pattern Indication
Following are the LED light pattern indications. RED LED indicates abnormal status
Button flashes every 1 second Button flashes every 2 seconds (if DHCP is configured) Button flashes every 4 seconds (if SIP server is configured) Button flashes every 6 seconds Button flashes briefly Red light steady. Ethernet link is down DHCP Failed or WAN No Cable HT486 fails to register Firmware Upgrading No STUN responses Device Malfunctions
Green LED indicates normal status
Button flashes every 2 second Button flashes every 1/10 second Button flashes every second Green light steady Message Waiting Indication Ringing Ringing interval In conversation

Configuration Guide

Configuring HandyTone-486 through Voice Prompt

DHCP Mode

Follow section 5.1 with voice menu option 01 to enable HandyTone-486 to use DHCP.

STATIC IP Mode

Follow section 5.1 with voice menu option 01 to enable HandyTone-486 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 Firmware Server IP Address
Follow section 5.1 with voice menu option 13 to configure the IP address of the firmware server.
6.1.4 Configuration Server IP Address
Follow section 5.1 with voice menu option 14 to configure the IP address of the configuration server.

6.1.5 Upgrade Protocol

Follow section 5.1 with voice menu option 15 to choose firmware and configuration upgrade protocol. User can choose between TFTP and HTTP.
6.1.6 Firmware Upgrade Mode
Follow section 5.1 with voice menu option 17 to choose firmware upgrade mode among the following three options: - always check - check when pre/suffix changes - never upgrade

6.1.7 WAN Port Web Access
Follow section 5.1 with voice menu option 12 to enable WAN Port Wed Access of the device configuration pages.
Configuring HandyTone-486 with Web Browser
HandyTone-486 has an embedded Web server that will respond to HTTP GET/POST requests. It also has embedded HTML pages that allow a user to configure the HandyTone-486 through a Web browser such as Microsofts IE and AOLs Netscape.
Access the Web Configuration Menu
HandyTone-486s web configuration page can be accessed via LAN or WAN port: From the LAN port: Directly connect a computer to the LAN port. Open a command window on the computer Type in ipconfig /release, the IP address etc. become 0.0.0.0. Type in ipconfig /renew, the computer gets an IP address in 192.168.2.1 by default Open a web browser, type in the default gateway IP address. You will see the login page of the device. http://192.168.2.1 From the WAN port:
The WAN port HTML configuration option is disabled by default from factory. To access the HTML configuration menu from the WAN port, first enable the WAN side HTTP access option via IVR option 12. With the WAN side HTTP access enabled, then get the WAN IP address of the HandyTone-486 through section 5.1 with menu option 02. The HandyTone-486s Web Configuration page can be accessed by the following URI via WAN port: http://HandyTone-IP-Address where the HandyTone-IP-Address is the WAN IP address of the HandyTone-486. NOTE: To type IP address into browser to get into the configuration page, please strip out the leading 0s 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.

End User Configuration

Once this request is entered and sent from a Web browser, the HandyTone-486 will respond with a login screen. 18
The password is case sensitive with a 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 SETTINGS configuration page. NOTE: If you can not log into the configuration page by using the default password, please check with your VoIP service provider. Most likely, the service provider has provisioned the device and configured for you and changed the default password.
After the correct password is entered into the login screen, the embedded Web server inside the device will respond with a BASIC SETTINGS configuration page.
End User Password Web Port IP Address
This field contains the password to access the Web Configuration Menu. The password is case sensitive with a maximum of 25 characters. This is the devices internal HTTP server port. Default is 80. There are 2 modes under which the HandyTone ATA can operate: - If DHCP mode is enabled, then all the field values 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 first DHCP server it discovers on the LAN it attaches to. To use PPPoE feature please set the PPPoE account settings if the HandyTone ATA is connected directly to a DSL modem. The HandyTone ATA will attempt to establish a PPPoE session if any of the PPPoE fields are set. - If Static IP mode is selected, then the IP address, Subnet Mask, Default Router IP address, DNS Server 1 (primary), DNS Server 2 (secondary) fields will need to be configured. These fields are set to zero by default.

DHCP hostname DHCP domain DHCP vendor class ID PPPoE account ID PPPoE password
This option specifies the name of the client. This field is optional but may be required by some Internet Service Providers. Default is blank. This option specifies the domain name that client should use when resolving hostnames via the Domain Name System. Default is blank. This option is used by clients and servers to exchange vendor-specific information. Default is blank. PPPoE username. Fill this field if your ISP requires you to use a PPPoE (Point to Point Protocol over Ethernet) connection. PPPoE account password.
Time Zone Daylight Savings Time
This parameters decides how the displayed date/time will be adjusted 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 Automatic Daylight Saving Time Rule shall have the following syntax: start-time;end-time;saving Both start-time and end-time have the same syntax: month,day,weekday,hour,minute month: 1,2,3,.,12 (for Jan, Feb,., Dec) day: [+|-]1,2,3,.,31 weekday: 1, 2, 3,., 7 (for Mon, Tue,., Sun), or 0 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 dayth iteration of the weekday (1st Sunday, 3rd Tuesday etc). If weekday us not zero and day is negative, then the daylight saving starts on the last dayth iteration of the weekday (last Sunday, 3rd last Tuesday etc). The saving is in the unit of minutes. The saving time may also be preceded by a negative (-) sign if subtraction is desired instead of addition. The default value for Automatic Daylight Saving Time Rule shall be set to 04,01,7,02,00;10,-1,7,02,00;60 which is the rule for US. Examples US/Canada where daylight saving time is applicable: 04,01,7,02,00;10,-1,7,02,00;60 This means the daylight saving time starts from the first Sunday of April at 2AM and ends the last Sunday of October at 2AM. The saving is 60 minutes (1hour).
PSTN Access Code Device Mode WAN Side HTTP Access
Default is *00. User can switch the phone to PSTN line connected to the Line port of ATA and make outgoing calls. Default is NAT router mode. HandyTone-486 Rev.2.0 can be configured in Bridge mode so the device functions as a bridge. Default is No. The access to configuration page via WAN port is disabled. This setting has no effect if the device is in Bridge mode. 20
Reply to ICMP on WAN port Cloned WAN MAC Address: LAN Subnet Mask
Unit will not respond to PING from WAN side if set to No. Allow user to set a specific MAC address. Set in Hex format. Sets the LAN subnet mask. Default value is 255.255.255.0. If bridge mode is selected, the LAN settings have no effect.

LAN DHCP Base IP: Base IP for the LAN port, which functions as a gateway for its LAN. Default value is 192.168.2.1 DHCP IP Lease Time DMZ IP: Port Forwarding: The amount of time that a given IP address will be valid for a LAN client. Value is set in units of hours. Default value is 120hr (5 Days). Forward all WAN IP traffic to a specific IP address if no matching port is used by HandyTone-486 itself or in the defined port forwarding. Allow user to forward a matching (TCP/UDP) port to a specific LAN IP address with a specific (TCP/UDP) port.
In addition to the Basic Settings configuration page, end users also have access to the device Status page. MAC Address WAN IP Address Product Model Software Version The device ID, in HEX format. troubleshooting. This is very important ID for ISP
This field shows WAN port IP address. This field contains the product model info, such as HT486 Rev:2.0 Program: This is the main software release. This number is always used for firmware upgrade. Current release is 1.0.8.32. Bootloader: current version is 1.0.8.11. HTML: current version 1.0.8.32. VOC: current version is 1.0.0.12 This shows how long the device has been up since the last reboot. This shows whether the unit is registered to service providers server or not. This field shows whether the PPPoE connection is up if the HandyTone ATA is connected to DSL modem. This shows what kind NAT the HandyTone ATA is connected to via its WAN port. It is based on STUN protocol. 21
System Uptime Registered PPPoE Link Up NAT
NAT Mapped IP NAT Mapped Port Other Statistical Status of ATA
WAN side public IP if connected to LAN of a SOHO router. WAN side SIP port if connected to LAN of a SOHO router. Self explainable, please see the page displayed.
Advanced User Configuration
To login to the Advanced User Configuration page, please follow the instructions in section 6.2.1 to get to the following login page. The password is case sensitive and the factory default password for Advanced User is admin. Advanced User configuration includes not only the end user configuration, but also advanced configuration such as SIP configuration, Codec selection, NAT Traversal Setting and other miscellaneous configuration.

Admin Password

Administrator password. Only administrator can configure the Advanced Settings page. Password field is purposely left blank for security reason after user clicks UPDATE button. This field is case sensitive and the maximum password length is 25 characters. IP address or Domain name provided by VoIP service provider IP address or Domain name of Outbound Proxy, or Media Gateway, or Session Border Controller. Used by ATA for firewall or NAT penetration in different network environment. If symmetric NAT is detected, STUN will not work and ONLY outbound proxy will provide solution for it. User account information, provided by VoIP service provider (ITSP), usually has the form of digit similar to phone number or actually a phone number. ID used for authentication, usually same as SIP user ID, but could be different and decided by ITSP. Account information, password for ATA to register to (SIP) servers of ITSP. SIP service subscribers name which will be used for Caller ID display. Local area code for North American Dial Plan.

SIP Server Outbound Proxy
SIP User ID Authenticate ID Authentication Password Name Home NPA

Preferred Vocoder

The HandyTone ATA supports 6 different codec types including G.711 A/U lawG.723.1, G.726, G.729A/B, iLBC. A user can configure Codecs in a preference list that will be included with the same preference order in SDP message. Encoding rate for G723 codec. By default, 6.3kbps rate is set. iLBC packet frame size. Default is 20ms. For Asterisk PBX, 30ms might be required.
G723 Rate: iLBC frame size:
iLBC payload type: Payload type for iLBC. Default value is 97. The valid range is between 96 and 127. Silence Suppression This enables/disables the silence suppression/VAD feature of G723. If set to Yes, when a silence is detected, small quantity of VAD packets (instead of audio packets) will be sent during this period. If set to No, this feature is disabled. Voice Frames per TX This field contains the number of voice frames transmitted in a single packet. When setting this value, the user should be aware of the requested packet time (used in SDP message) as a result of configuring this parameter. This parameter is associated with the first codec in the above codec Preference List or the actual used payload type negotiated between the 2 conversation parties at run time. e.g., if the first codec is configured as G723 and the Voice Frames per TX is set to be 2, then the ptime value in the SDP message of an INVITE request will be 60ms because each G723 voice frame contains 30ms of audio. Similarly, if this field is set to be 2 and if the first codec chosen is G729 or G711 or G726, then the ptime value in the SDP message of an INVITE request will be 20ms. If the configured voice frames per TX exceeds the maximum allowed value, the HandyTone ATA will use and save the maximum allowed value for the corresponding first codec choice. The maximum value for PCM is 10(x10ms) frames. For G726, it is 20 (x10ms) frames. And 32 (x30ms) and 64 (x10ms) frames for G723 and G729 respectively. T.38 (Auto Detect) FoIP by default, or Fax Pass-Through. This field defines the layer 3 QoS parameter which can be the value used for IP Precedence or Diff-Serv or MPLS. Default value is 48. Layer 2 QoS settings for VoIP traffic. Default setting is blank. VLAN supported equipment is required if user needs to change these settings. Layer 2 QoS settings for LAN port device traffic. Default setting is blank. VLAN supported equipment is required if user needs to change these settings.

Keep-alive interval Use NAT IP: Use STUN keepalive to detect networks connectivity Proxy-Require
Subscribe for MWI: Default is No. When set to Yes a SUBSCRIBE for Message Waiting Indication will be sent periodically. Offhook Auto-Dial This parameter allows a user to configure a User ID or extension number to be automatically dialed upon offhook. Please note that only the user part of a SIP address needs to be entered here. The HandyTone ATA will automatically append the @ and the host portion of the corresponding SIP address. Note: Please write down the IP address of the ATA if you use this feature as it will prevent you to access the IVR and the only way to access the device configuration is via the web configuration page. Default is Yes. locally. Advanced call features using start codes are supported
Enable Call Features Use Bell-style 3-way Conference Disable Call Waiting Disable CallWaiting Caller-ID
If this parameter is set to Yes, user will be able to make Bellcore style 3-way conference. *23 will be disabled. Default is No. User can use star codes to enable/disable call waiting. Default is No.

Send DTMF

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. Multiple selections of DTMF method are supported. This parameter sets the payload type for DTMF using RFC2833 Default is No. If set to Yes, flash will be sent as a DTMF event. The amount of time the hookflash is pressed that will cause the device to onhook. Default is 800ms. Selects the impedance of the analog telephone connected to the Phone port. Select the Caller ID Scheme to suit the standard of different area. Bellcore (North America) CID-Canada DTMF-Brazil DTMF-Sewden DTMF (Denmark) ETSI-DTMF (Finland, Sweden) ETSI-FSK (France, Germany, Norway, Taiwan, UK-CCA) The onhook voltage can be selected according to the line voltage depending on the analog phone used. The low power/high power will increase/decrease the output current. Selecting "low power" will make loop current limit = 20mA, selecting "high power" make loop current limit = 32mA. The default selection is 36V (High Power). Select Polarity Reversal to adapt some call charge/billing system. Default is No. This is the URI or IP address of the NTP (Network Time Protocol) server, which the HandyTone ATA will use to synchronize the date/time. If this parameter is set to Yes, user ID will be sent as anonymous, essentially blocks the Caller ID from displaying.

DTMF Payload Type Send Flash Event Onhook Threshold FXS Impedance Caller ID Scheme

Onhook Voltage

Polarity Reversal NTP server Send Anonymous
Anonymous Method 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. Time to Ring Special Features The duration of ringing when a call is not answered. Default is 60 seconds. Default is Standard. Choose the selection to meet some special requirements from Soft Switch vendors like Nortel, Broadsoft, etc.
CBCOM Encode CBCOM Encoder 1.1 Key Syslog Server Syslog Level
SIP, RT(C)P and T.38 modes, 1.0 and 1.1 Key to be used by CBCOM The IP address or URL of System log server. This feature is especially useful for ITSP (Internet Telephone Service Provider). 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: product model/version on boot up (INFO level) NAT related info (INFO level) sent or received SIP message (DEBUG level) SIP message summary (INFO level) inbound and outbound calls (INFO level) registration status change (INFO level) negotiated codec (INFO level) Ethernet link up (INFO level) SLIC chip exception (WARNING and ERROR levels) 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 [00:0b:82:00:a1:be][000] Ethernet link is up Session Expiration 192.168.1.14 GS_LOG:
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.

Min-SE

The minimum session expiration (in seconds). The default value is 90 seconds.
Caller Request Timer Callee Request Timer Force Timer
If selecting Yes the phone will use session timer when it makes outbound calls if remote party supports session timer. If selecting Yes the phone will use session timer when it receives inbound calls with session timer request. If selecting Yes the phone will use session timer even if the remote party does not support this feature. Selecting No will allow the phone to enable session timer only when the remote party support this feature. To turn off Session Timer, select No for Caller Request Timer, Callee Request Timer, and Force Timer. As a Caller, select UAC to use the phone as the refresher, or UAS to use the Callee or proxy server as the refresher. As a Callee, select UAC to use caller or proxy server as the refresher, or UAS to use the phone as the refresher. Session Timer can be refreshed using INVITE method or UPDATE method. Select Yes to use INVITE method to refresh the session timer. Default method is HTTP. Firmware upgrade may take up to 10 minutes depending 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 is blank. If configured, HT486 rev. 2.0 will request the firmware file with the prefix. This setting is useful for ITSPs. End user should keep it blank. Default is blank. End user should keep it blank. Default is blank. End user should keep it blank. Default is blank. End user should keep it blank.

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 Choose Yes to enable automatic upgrade and provisioning and input the number, in minutes, you want the HT to check for an update. When set to No, HT502 will only do upgrade once at boot up. Always check for New Firmware at Boot up will check for new firmware every time the device reboots. Check New Firmware only when F/W pre/suffix changes will check for updates only when the pre/suffix has been changed. Firmware Key For firmware encryption. It should be 32 digit in Hexadecimal Representation. End user should keep it blank.
Authenticate Conf File Lock keypad update Allow conf SIP Account in Basic Settings Volume Amplification Powerline Ring Tone
Default is No. If set to Yes, configuration file would be authenticated before acceptance. 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. Default No. If set to Yes, user ID, authentication IP, authentication password and display name can be configured in BASIC SETTINGS page. Handset volume adjustment. RX is for receiving volume, TX is for transmission volume. Default values are 0dB for both parameters. +6dB generates the highest volume and -6dB generates the lowest volume. This setting allows user to configure the ringing frequencies and cadences.
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 period of ringing (On time in ms) while OFF is the period of silence. In order to set a continuous ring, OFF should be zero. Otherwise it will ring ON ms and a pause of OFF ms and then repeat the pattern. Up to three cadences are supported. Disable Line Echo Canceller (LEC): Default is No. If set to Yes, echo canceller is not used.
Saving the Configuration Changes
Once a change is made, users should click on the Update button in the Configuration page. The HandyTone-486 will then display a screen to confirm that the changes have been saved.
Rebooting the HandyTone-486 from remote
The user/administrator of the HandyTone-486 can remotely reboot the HandyTone-486 by pressing the Reboot button at the bottom of the configuration page. Once done, a screen will be displayed to indicate that rebooting is underway.
Configuration through a Central Server
Grandstream HandyTone ATAs can be automatically configured from a central provisioning system. 29
When HandyTone ATA boot up, it will send TFTP or HTTP request to download configuration file, cfg000b82xxxxxx, where 000b82xxxxxx is the MAC address of the HandyTone ATA. The configuration files can be downloaded via TFTP or HTTP from the central server. A service provider or an enterprise with large deployment of HandyTone ATA can easily manage the configuration and service provisioning of individual devices remotely from a central server. Grandstream has a provisioning system called GAPS (Grandstream Automated Provisioning System) that is used to support automated configuration of Grandstream devices. GAPS uses enhanced (NAT friendly) TFTP or HTTP (thus no NAT issues) and other communication protocols to communicate with each individual Grandstream device for firmware upgrade, remote reboot, etc. Grandstream provides GAPS service to VoIP service providers. Use GAPS for either simple redirection or with certain special provisioning settings. At boot-up, Grandstream devices by default point to Grandstream provisioning server GAPS, based on the unique MAC address of each device, GAPS provision the devices with redirection settings so that they will be redirected to customers TFTP or HTTP/HTTPS server for further provisioning. Grandstream also provide GAPSLite software package which contains our NAT friendly TFTP server and a configuration tool to facilitate the task of generating device configuration files. The GAPSLite configuration tool is now free to end users. The tool and configuration template are available for download from http://www.grandstream.com/configurationtool.html

Managing Firmware and Configuration File Download
When Automatic Upgrade is set to Yes, Service Provider can use P193 (Auto Check Interval, in minutes, default and minimum is 60 minutes) to have the devices periodically check with either Firmware Server or Config Server, whenever they are defined. This allows the device periodically check if there are any new changes need to be taken on a scheduled time. By defining different intervals in P193 for different devices, Server Provider can spread the Firmware or Configuration File download in minutes to reduce the Firmware or Provisioning Server load at any given time.

Warning !!!

Restore Factory Default Setting
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: a. b. c. d. 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.
The device will reboot automatically and restore to factory default setting.
NOTE: Please be aware by default the HandyTone-486 WAN side HTTP access is disabled. After a factory reset, the devices web configuration page can be accessed only from its LAN port, please refer to instructions in section 6.2.1 for details. 34