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Grandstream HT502

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Grandstream HT502Grandstream HandyTone 502 - VoIP phone adapter

External, AC 120/230 V, 2 ports

The Handytone HT-502 next generation analog telephone adapter based on SIP standards represents a powerful, affordable and high quality manageable IP telephony solution for home or office applications.

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Reproduction or transmittal of the entire or any part, in any form or by any means, electronic or print, for any purpose is not permitted without the express written permission of Grandstream Networks, Inc.
Page 4 of 31 Last Updated: 12/2008

CONNECT YOUR HT502

Equipment Packaging
The HT502 ATA package contains: One HT502 Main Case One Universal Power Adaptor One Ethernet Cable One HT502 Vertical Stand

Connecting the HT502

The HT502 is designed for easy configuration and easy installation. Configure the HT502 following the directions in the Configuration section of this manual. 1. Connect a standard touch-tone analog telephone to the PHONE port. 2. Insert a standard RJ11 telephone cable into the Phone1 port and connect the other end of the telephone cable to the analog telephone. 3. Insert the Ethernet cable into the WAN port of HT502 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 HT502 if it is being used as a router. 5. Insert the power adapter into the HT502 and connect it to a wall outlet. The HT502 Analog Telephone Adaptor is an all-in-one VoIP integrated device designed to be a total solution for networks providing VoIP services. The HT502 VoIP features and functions are available using a regular analog telephone. Figure 1: Connecting the HT502

HT-502 Front View

HT-502 Back View

Display LEDs (green)

RJ-45 Ports 10/100 Mbps

Power Supply (12V)

RJ-11 FXS Ports (Phone)
The HT502 has 2 FXS port. Both FXS ports can have a separate SIP account. This is a key feature of HT502 as it supports simultaneous calls on both FXS ports.
Page 5 of 31 Last Updated: 12/2008
TABLE 1: DEFINITIONS OF THE HT502 CONNECTORS Power Cable WAN Port (RJ-45) LAN Port (RJ-45) RESET PHONE1 (RJ-11) PHONE2 (RJ-11) Power adapter connection Connect to the internal LAN network or router. Connect the LAN port with an Ethernet cable to your PC. Factory Reset button. Press for 7 seconds to reset factory default settings. FXS port to be connected to analog phones / fax machines. FXS port to be connected to analog phones / fax machines.
There are five (5) LED buttons that help you manage the status of your HandyTone. TABLE 2: DEFINITIONS OF THE HT502 LEDS
LEDs POWER LED WAN LED LAN LED PHONE1 / PHONE2 LED Indicates Power. Remains ON when power is connected Indicates LAN (or WAN) port activity Indicates PC (or LAN) port activity Indicate status of the respective FXS Ports-PHONE1 / PHONE2 on the back panel Busy ON (Solid Green) Available OFF Slow blinking FXS LEDs indicates voicemail for that port.

NOTE: All LEDs display green when ON. Slow blinking of WAN and LAN LED together indicate the product in firmware upgrading or provision state. Figure 2: HT502 Connection Diagram
Internet ADSL/Cable Modem Ethernet Analog Phone

Analog Phone

Cordless Phone
Page 6 of 31 Last Updated: 12/2008

PRODUCT OVERVIEW

The HT502 is a full feature voice and fax-over IP device that offers a high-level of integration including dual 10M/100Mbps network ports with integrated router, NAT, DHCP server, dual port FXS telephone gateway, market-leading sound quality, rich functionalities, and a compact and lightweight design. The VoIP network signaling protocol supported is SIP. The HT502 fully compatible with SIP industry standard and can interoperate with many other SIP compliant devices and software on the market. Moreover, it supports comprehensive voice codecs including G.711 (a/-law), G.723.1, G.726 (16/24/32/40 bit rates), AAL2 (all G.726), G.729A/B/E and iLBC.
Software Features Overview
2 FXS ports Two RJ-45 ports (switched or routed) 2 SIP accounts & profiles Supports Voice Codecs: G711(a/, Annex I & II), G723.1A, G726 (ADPCM with 16/24/32/40 bit rates), G729 A/B/E, iLBC T.38 Fax Comprehensive Dial Plan support for Outgoing calls. G.168 Echo Cancellation Voice Activation Detection (VAD), Comfort Noise Generation (CNG), and Packet Loss Concealment (PLC) Supports PSTN/PBX analog telephone sets or analog trunks
TABLE 3: HT502 TECHNICAL SPECIFICATIONS
Telephone Interfaces Network Interface LED Indicators Reset Button Voice over Packet Capabilities 2 FXS ports, 2 SIP accounts Two (2) 10M/100 Mbps, RJ-45 Power, WAN, LAN, PHONE1 and PHONE2 Factory Reset button. Voice Activity Detection (VAD) with CNG (comfort noise generation) and PLC (packet loss concealment), Dynamic Jitter Buffer, Modem detection & auto-switch to G.711, Packetized Voice Protocol Unit (supports RTP/RTCP and AAL2 protocol), G.168 compliant Echo Cancellation, LEC (line echo cancellation) with NLP, Asymmetric RTP stream G.711 + Annex I (PLC), Annex II (VAD/CNG format) encoder and decoder, G.723.1A, G.726(ADPCM), G.729A/B/E, iLBC, G.726 provides proprietary VAD, CNG, and signal power estimation, Voice Play Out unit (reordering, fixed and adaptive jitter buffer, clock synchronization), AGC (automatic gain control), Status output, Decoder controlling via voice packet header Yes, NAT Router or Switched Mode Yes T.38 compliant Group 3 Fax Relay up to 14.4kpbs and auto-switch to G.711 for Fax Pass-through, Fax Datapump V.17, V.19, V.27ter, V.29 for T.38 fax relay Diffserve, TOS, 802.1 P/Q VLAN tagging RTP/RTCP Flexible DTMF transmission method, user interface of In-audio, RFC2833, and/or SIP Info HT-502 User Manual Firmware Version 1.0.1.21 Page 7 of 31 Last Updated: 12/2008

Voice Compression

DHCP Server/Client Telnet Server Fax over IP QoS IP Transport DTMF Method
IP Signaling Provisioning Control Management
SIP (RFC 3261) TFTP, HTTP, HTTPS TLS/SIPS , SIP over TCP/TLS Syslog support, HTTPS and telnet, remote management using Web browser, Support Layer 2 (802.1Q, VLAN, 802.1p) and Layer 3 QoS (Tos, DiffSery, MPLS), Auto/manual provisioning system Yes Yes Output: 12VDC / Input: 100240 VAC/50-60 Hz Operational: F or C Storage: F / Humidity: 1090% Non-condensing

o o o o o o

Dial Plan UPnP Power Environmental
Dimensions (H x W x D) Short and long haul Call Handling Features
115mm (L) x 75mm (W) x 27mm (H) REN3: Up to150 ft on 24 AWG line Caller ID display or block, Call waiting caller ID, Call waiting/flash, Call transfer, hold, forward, mute, 3-way conferencing, message waiting, Do-Not-Disturb (DND), call-return service Bellcore Type 1 & 2, ETSI, BT, NTT, and DTMF-based CID Yes EN55022/EN55024 and FCC part15 Class B UL
Caller ID Polarity Reversal / Wink EMC Safety

Hardware Specification

The table below lists the hardware specification of HT502. TABLE 4: HT502 HARDWARE SPECIFICATION
LAN Interface LED Universal Switching Power Adaptor 2 x RJ45 10/100Mbps (integrated router) 5 LEDs (GREEN) Input: 100-240V AC, 50/60Hz, 0.5A Max Output: 12V DC, 1.25A UL certified Dimension Weight Temperature Humidity Compliance 115mm (L) x 75mm (W) x 27mm (H) 94 g (0.21lbs) 32~104F / 0~40C 10% - 90% (non-condensing)
Page 8 of 31 Last Updated: 12/2008

BASIC OPERATIONS

Understanding HT502 Voice Prompt
HT502 has a built-in voice prompt menu for simple device configuration. The IVR menu and the LED button work with any of the FXS port. Pick up the handset and dial *** to use the IVR menu. TABLE 5: HT502 IVR MENU DEFINITIONS
Menu Main Menu Voice Prompt Enter a Menu Option Options Press * for the next menu option Press # to return to the main menu Enter 01-05, 07,10, 12-17,47 or 99 menu options Press 9 to toggle the selection If using Static IP Mode, configure the IP address information using menus 02 to 05. If using Dynamic IP Mode, all IP address information comes from the DHCP server automatically after reboot. The current WAN IP address is announced If using Static IP Mode, enter 12 digit new IP address. Same as menu 02 Same as menu 02 Same as menu 02 Press 9 to move to the next selection in the list: PCM U / PCM A iLBC G-726 G-723 G-729 Announces the Mac address of the unit. Press 9 to toggle between enable / disable Announces current Firmware Server IP address. Enter 12 digit new IP address. Announces current Config Server Path IP address. Enter 12 digit new IP address. Upgrade protocol for firmware and configuration update. Press 9 to toggle between TFTP / HTTP / HTTPS Firmware version information. Firmware upgrade mode. Press 9 to toggle among the following three options: - always check - check when pre/suffix changes - never upgrade Enter the target IP address to make a direct IP call, after dial tone. (See Make a Direct IP Call.) User can make internal calls between two FXS ports on the same HT502 even without being registered to SIP server. By dialing *** and 70 user can reach the other FXS port Number of Voice Mails Press 9 to reboot the device Enter MAC address to restore factory default setting HT-502 User Manual Firmware Version 1.0.1.21 Page 9 of 31 Last Updated: 12/2008

Page 15 of 31 Last Updated: 12/2008
NOTE: If you can not log into the configuration page by using the default password, please check with the VoIP service provider. It is most likely the VoIP service provider has provisioned the device and configured for you therefore the password has already been changed.

Important Settings

The end-user must configure the following settings according to the local environment. settings on the web configuration pages are set to the default values. NOTE: Most
NAT SETTINGS If you plan to keep the gateway within a private network behind a firewall, we recommend using STUN Server. The following three (3) settings are useful in the STUN Server scenario: 1. STUN Server (under Advanced Settings webpage) Enter a STUN Server IP (or FQDN) that you may have, or look up a free public STUN Server on the internet and enter it on this field. If using Public IP, keep this field blank. 2. Use Random Ports (under Advanced Settings webpage) This setting depends on your network settings. Generally if you have multiple IP devices under the same network, it should be set to Yes. If using a Public IP address, set this parameter to No. 3. NAT Traversal (under the Profile web pages) Set this to Yes when gateway is behind firewall on a private network. DTMF METHODS DTMF Settings are in Profile pages. DTMF in-audio DTMF via RTP (RFC2833) DTMF via SIP INFO
Enable one or more DTMF methods based on your PBX system. PREFERRED VOCODER (CODEC) The HT502 supports a broad range of voice codecs. Under Profile web pages, choose your preferred order of different codecs: PCMU/A (or G711/a) G729 A/B/E G723 G726 (16/24/32/40) iLBC AAL2 (all G.726)
Page 16 of 31 Last Updated: 12/2008

TABLE 7: BASIC SETTINGS

End User Password Password to access the Web Configuration Menu. This field is case sensitive with a maximum length of 25 characters. By default, HTTP uses port 80. This field is for customizable web port. Default is set to YES. There are two modes to operate the HT502: DHCP mode: all the field values for the Static IP mode are not used (even though they are still saved in the Flash memory.) The HT502 acquires its IP address from the first DHCP server it discovers from the LAN it is connected. Using the PPPoE feature: set the PPPoE account settings. The HT502 will establish a PPPoE session if any of the PPPoE fields is set. Static IP mode: configure the IP address, Subnet Mask, Default Router IP address, DNS Server 1 (primary), DNS Server 2 (secondary) fields. These fields are set to zero by default. Default is blank. 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 HT500. Used by clients and servers to exchange vendor-specific information. PPPoE username. Necessary if ISP requires you to use a PPPoE (Point to Point Protocol over Ethernet) connection. PPPoE account password. Default is blank. This field is optional. If your ISP uses a service name for the PPPoE connection, enter the service name here. Controls how the date/time is displayed according to the specified time zone. The syntax is std offset dst [offset],start[/time],end[/time] Default is set to : MTZ+6MDT+5,M3.2.0,M11.1.0 MTZ+6MDT+5, Time zone with 6 hours offset with 1 hour ahead which is the US central time. It is positive (+) if the local time zone is west of the Prime Meridian and negative (-) if it is east. Prime Meridian (a.k.a: International or Greenwich Meridian) M3.2.0,M11.1.0 st The 1 number indicates Month: 1,2,3,.,12 (for Jan, Feb,., Dec) nd th The 2 number indicates the n iteration of the weekday: (1st Sunday, 3rd Tuesday etc) rd The 3 number indicates Weekday: 0,1, 2,.,6(for Sun, Mon, Tue,., Sat) Therefore, this example is the DST which starts from the second Sunday of March to the 1st Sunday of November. Language Languages supported with voice prompt and web interface, except Spanish that it is only in IVR. This parameter controls whether the device is working in NAT router mode or Bridge mode. Save the setting and reboot prior to configuring HT502.

Web Port Telnet Server IP Address

DHCP hostname

DHCP domain
DHCP vendor class ID PPPoE account ID
PPPoE password PPPoE Service Name
Time Zone Self Defined Time Zone

Device Mode

Page 17 of 31 Last Updated: 12/2008

NAT Maximum Ports

The number of ports that can be managed while in NAT router mode. Range: 0 4096, default is 1024. Typically one port per connection. NAT TCP idle timeout in seconds. Connection will be closed after preconfigured, timeout if not refreshed. Range: 0 - 3600 NAT TCP idle timeout in seconds. Connection will be closed after preconfigured, timeout if not refreshed. Range: 0 3600, default is 300 The maximum uplink bandwidth permitted by the device. This function is disabled by default. The total bandwidth can be set as: 128K, 256K, 512K, 1M, 4M or 10M. The primary function of this setting is to reserve bandwidth for VoIP. Example: if 64 is configured, there will be at least 64kbps reserved for VoIP. The maximum downlink bandwidth permitted by the device. This function is disabled by default. The total bandwidth can be set as: 128K, 256K, 512K, 1M, 4M or 10M. The primary function of this setting is to reserve bandwidth for VoIP. Example: if 128 is configured, there will be at least 128kbps reserved for VoIP. When set to Yes, the HT502 acts as an UPnP gateway for your UPnP enabled applications. UPnP = Universal Plug and Play Default is No. When set to Yes, the HT502 responds to the PING command from other computers, but is also made vulnerable to DOS attacks. Default is No. When set to Yes, the user can access the web configuration pages through the WAN port, instead of through the PC port. Warning: this configuration is less secure than the default option. This allows the user to change/set a specific MAC address on the WAN interface. Note: Set in Hex format When set to Yes, device will function as a simple router and LAN port will provide IP addresses to internal network. Connect the WAN port to ADSL/Cable modem or any other equipment that provides access to public Internet. Base IP Address for a LAN port. Default factory setting is 192.168.2.1 Default value is 100. The last segment of IP address assigned to the HT50x in the LAN Network. Default configuration assigns IP address (to local network devices) starting from 192.168.2.100. Default value is 199. This parameter allows a user to limit the number of local network devices connected to the internal router. Default configuration assigns IP address (to devices connected to the LAN port) in a range from 192.168.2.100 up to 192.168.2.199. Sets the LAN subnet mask. Default value is 255.255.255.0 Base IP address for the LAN port, which functions as default gateway for its LAN. Default value is 192.168.2.1 Default value is 120 hrs (5 days). The length of time the IP address is assigned to the LAN clients. Value is set in units of hours. This function forwards all WAN IP traffic to a specific IP address if no matching port is used by HT502 or in the defined port forwarding. Forwards a matching (TCP/UDP) port to a specific LAN IP address with a specific (TCP/UDP) port.

NAT TCP Timeout

NAT UDP Timeout

Uplink Bandwidth

Downlink Bandwidth

Enable UPnP

Reply to ICMP on WAN Port WAN Side HTTP/Telnet Access
Cloned WAN MAC Address: Enable LAN DHCP
LAN DHCP Base IP LAN DHCP Start IP

LAN DHCP End IP

LAN Subnet Mask LAN DHCP Base IP:

DHCP IP Lease Time

DMZ IP:

Port Forwarding

In addition to the Basic Settings configuration page, end users also have access to the Device Status page.
Grandstream Networks, Inc. HT-502 User Manual Firmware Version 1.0.1.21 Page 18 of 31 Last Updated: 12/2008

TABLE 8: STATUS PAGE

MAC Address The device ID, in HEX format. This is very important ID for ISP troubleshooting. LAN and WAN Mac addresses will appear in this place. The LAN MAC address will be used for provisioning and can be found on the label coming with original box and on the label located on the bottom panel of the device. This field shows WAN IP address of HT502. This field contains the product model info. Program: This is the main software release. This number is always used for firmware upgrade. Current release is 1.0.1.21 Boot and Loader are seldom changed. Bootloader: current version is 1.0.0.9. Core: current version 1.0.0.25 Base: current version is 1.0.0.76 Shows system up time since the last reboot. Indicates whether the PPPoE connection is up if the HT502 is connected to DSL modem. This filed indicates the type of NAT connection used by the HT502 via its WAN port. Displays relevant information regarding the each FXS port. Port FXS1 FXS2 Hook On Hook On Hook Registration Registered Registered DND Yes No Forward Busy Forward Delayed Forward
WAN IP Address Product Model Software Version
System Up Time PPPoE Link Up NAT Port Status
Both FXS port1 and FXS port2 are registered with this SIP Server. FXS Port 1 user has set Do Not Disturb. FXS Port 1 user has set his calls to be forwarded unconditionally to ext 613. FXS Port 2 user has set his calls to forward to 614 when his phone is busy.

SIP transport NAT Traversal (STUN)
-A Record (for resolving IP Address of target according to domain name) -SRV (DNS SRV resource records indicates how to find services for various protocols) -NAPTR/SRV (Naming Authority Pointer according to RFC 2915) One mode can be chosen for the client to look up server. The default value is A Record User ID is Phone Number SIP Registration Unregister on Reboot Outgoing Call w/o Registration Register Expiration If the HT502 has an assigned PSTN telephone number, this field should be set to Yes. Otherwise, set to No. If set to Yes, a user=phone parameter will be appended to the From header in SIP request. Controls whether the HT502 needs to send REGISTER messages to the proxy server. The default setting is Yes. Default is No. If set to Yes, the SIP users registration information will be cleared on reboot. Default is No. If set to Yes, user can place outgoing calls even when not registered (if allowed by ITSP) but is unable to receive incoming calls. This parameter allows the user to specify the time frequency (in minutes) the HT502 refreshes its registration with the specified registrar. The default interval is 60 minutes (or 1 hour). The maximum interval is 65535 minutes (about 45 days). Retry registration if the process failed. Default is 30 seconds. Defines the local SIP port the HT502 will listen and transmit. The default value for FXS port 1 is 5060. The default value for FXS port 2 is 5062. Defines the local RTP-RTCP port pair the HT502 will listen and transmit. It is the base RTP port for channel 0. When configured, channel 0 uses this port _value for RTP and the port_value+1 for its RTCP; channel 1 uses port_value+2 for RTP and port_value+3 for its RTCP. The default value for FXS port 1 is 5004. The default value for FXS port 2 is 5012. Use Random Port Refer to Use Target Contact Transfer on Conference Hang up This parameter forces the random generation of both the local SIP and RTP ports when set to Yes. This is usually necessary when multiple HT502 are behind the same NAT. Default is NO. If set to YES, then for Attended Transfer, the Refer-To header uses the transferred targets Contact header information. Default is No. In which case if the conference originator hangs up the conference will be terminated. When option YES is chosen, originator will transfer other parties to

Registration Retry Wait Time Local SIP port Local RTP port
each other so that B and C can choose either to continue the conversation or hang up.
Remove OBP from Route Header Support SIP Instance ID Validate incoming SIP message Check SIP Incoming UserID SIP T1 Timeout SIP T2 Interval DTMF Payload Type DTMF in-audio DTMF via RFC2833 DTMF via SIP INFO Grandstream Networks, Inc. Default is No. When option YES is chosen, the Out Bound Proxy will be removed from Route header. Default is Yes. If set to Yes, the contact header in REGISTER request will contain SIP Instance ID as defined in IETF SIP Outbound draft. Default is No. If set to yes all incoming SIP messages will be strictly validated according to RFC rules. If message will not pass validation process, call will be rejected. Default is No. Check the SIP User ID in Request URI. If they dont match, the call will be rejected. T1 is an estimate of the round-trip time between the client and server transactions. If the network latency is high, select larger value for more reliable usage. Maximum retransmission interval for non-INVITE requests and INVITE responses. Sets the payload type for DTMF using RFC2833. Send DTMF as inband (in-audio). Send DTMF via RTP (According to RFC 2833). Send DTMF via SIP INFO message. HT-502 User Manual Firmware Version 1.0.1.21 Page 23 of 31 Last Updated: 12/2008
Send Flash Event Enable Call Features Offhook Auto-Dial
Default is No. If set to yes, flash will be sent as DTMF event. Default is Yes. (If Yes, call features using star codes will be supported locally) This parameter allows users to configure a User ID or extension number that is automatically dialed when off-hook. Only the user part of a SIP address needs is entered here. The HT502 will automatically append the @ and the host portion of the corresponding SIP address. SIP Extension to notify SIP server that the unit is behind the NAT/Firewall. NAT IP address used in SIP/SDP message. Default is blank. Custom Ring Tone 1 to 3 with associate Caller ID: when selected, if Caller ID is configured, then the device will ONLY uses this ring tone when the incoming call is from the Caller ID. System Ring Tone is used for all other calls. When selected but no Caller ID is configured, the selected ring tone will be used for all incoming calls. Distinctive ring tones can be configured not only for matching a whole number, but also for matching prefixes. In this case symbol * (star) will be used. If server supports Alert-Info header and standard ring tone set (Bellcore) or distinctive ring tone 1-10 is specified, then the ring tone in the Alert-Info header from server will be used. For example: if configured as *617, Ring Tone 1 will be used in case of call arrived from the area code 617. Any other incoming call will ring using cadence defined in parameter System Ring Cadence located under Advanced Settings Configuration page.

Proxy-Require Use NAT IP Distinctive Ring Tone
Disable Call Waiting Disable Call-Waiting Caller ID Disable Call Waiting Tone Disable Visual MWI Ring Timeout Delayed Call Forward Wait Time No Key Entry Timeout Early Dial
Default is No. If set to YES Call Waiting indication information will not be provided to analog phone connected to this FXS port. Default is No. If set to YES Call Waiting caller ID will not be provided to analog phone connected to this FXS port. Default is No. This is to disable the stutter Call Waiting Tone when a Call Waiting information arrives. The CWCID information will still be displayed. If set to Yes, the MWI information will not be transferred to the analog phone connected to the FXS port. Incoming call will stop ringing when not picked up given a specific period of time. Default value is 20 seconds. In case this feature activated using * codes (*92 code), the call will be forwarded after this preconfigured amount of time. Default is 4 seconds. Dialing process is completed and outgoing call is initiated if no key entry occurs during this preconfigured interval. Default is No. Use only if proxy supports 484 response. This parameter controls whether the phone will send an early INVITE each time a key is pressed when a user dials a number. If set to Yes, an INVITE is sent using the dial-number collected thus far; Otherwise, no INVITE is sent until the (Re-)Dial button is pressed or after about 5 seconds have elapsed if the user forgets to press the Re-Dial button. The Yes option should be used ONLY if there is a SIP proxy configured and the proxy server supports 484 Incomplete Address response. Otherwise, the call will likely be rejected by the proxy (with a 404 Not Found error). This feature does NOT work with and should NOT be enabled for direct IP-to-IP calling. Sets the prefix added to each dialed number. Allows users to configure the # key as the Send (or Dial) key. If set to Yes, # will send the number. In this case, this key is essentially equivalent to the Dial key. If set to No, this # key can be included as part of number.
Dial Plan Prefix Use # as Dial Key
Page 24 of 31 Last Updated: 12/2008

Dial Plan

Dial Plan Rules: 1. Accept Digits: 1,2,3,4,5,6,7,8,9,0 , *, #, A,a,B,b,C,c,D,d 2. Grammar: x - any digit from 0-9; xx+ - at least 2 digits number; xx. at least 1 digit number. ^ - exclude; [3-5] - any digit of 3, 4, or 5; [147] - any digit 1, 4, or 7; <2=011> - replace digit 2 with 011 when dialing | - or Example 1: {[369]11 | 1617xxxxxxx} Allow 311, 611, 911, and any 11 digit numbers with leading digits 1617 Example 2: {^1900x+ | <=1617>xxxxxxx} Block any number of leading digits 1900 and add prefix 1617 for any dialed 7 digit numbers Example 3: {1xxx[2-9]xxxxxx | <2=011>x+} Allow any combinations of numbers with 11 digits which has a leading digit 1, but 5th digit cannot be 0 or 1. Or any length of numbers with a minimum of 2 digits beginning with 2, with the leading digit replaced with 011. 3. Default: Outgoing - {x+} Example of a simple dial plan used in a Home/Office in the US: { ^1900x. | <=1617>[2-9]xxxxxx | 1[2-9]xx[2-9]xxxxxx | 011[2-9]x. | [3469]11 } Explanation of example rule (reading from left to right): ^1900x. - prevents dialing any number started with 1900 <=1617>[2-9]xxxxxx - allows dialing to local area code (617) numbers by dialing 7 numbers and 1617 area code will be added automatically 1[2-9]xx[2-9]xxxxxx |- allows dialing to any US/Canada Number with 11 digits length 011[2-9]x. - allows international calls starting with 011 [3469]11 - allow dialing special and emergency numbers 311, 411, 611 and 911 Note: In some cases user wishes to dial strings such as *123 to activate voice mail or other application provided by service provider. In this case * should be predefined inside dial plan feature and the Dial Plan should be: { *x+ }.

Fax Mode Fax Tone Detection Mode Jitter Buffer Type Jitter Buffer Length SRTP Mode SLIC Setting Caller ID Scheme Polarity Reversal

Loop Current Disconnect

Loop Current Disconnect Duration Hook Flash Timing
Here can be configured duration of such voltage drop described in topic above. Time period when the cradle is pressed (Hook Flash) to simulate FLASH. To prevent unwanted activation of the Flash/Hold and automatic phone ring-back, adjust this time value. On-hook timing is the minimum time for an on-hook event to be validated. Voice path volume adjustment. Rx is a gain level for signals transmitted by FXS Tx is a gain level for signals received by FXS.

On Hook Timing Gain

Default = 0dB for both parameters. Loudest volume: +6dB Lowest volume: -6dB. User can adjust volume of call on either end using the Rx Gain Level parameter and the Tx Gain Level parameter located on the FXS Port Configuration page. If call volume is too low when using the FXS port (ie. the ATA is at user site), adjust volume using the Rx Gain Level parameter under the FXS Port Configuration page. If voice volume is too low at the other end, user may increase the far end volume using the Tx Gain Level parameter under the FXS Port Configuration page. This function lets you configure ring tone cadence preferences. User has 10 choices. The configuration, completed in Distinctive Ring Tones block in the same page, applies to ring tones cadences configured here.

Ring Tones

Saving the Configuration Changes
Click the Update button in the Configuration page to save the changes to the HT502 configuration. The following screen confirms that the changes are saved. Reboot or power cycle the HT502 to make the changes take effect.
Rebooting the HT502 from Remote
Remotely reboot the HT502 by clicking the Reboot button at the bottom of the configuration page. When finished, re-login to the HT502 after waiting for about 30 seconds.
Configuration through a Central Server
Grandstream HT502 can be automatically configured from a central provisioning system. When HT502 boot up, it will send TFTP or HTTP/HTTPS request to download configuration file, cfg000b82xxxxxx, where 000b82xxxxxx is the LAN MAC address of the HT502. The configuration file can be downloaded via TFTP or HTTP/HTTPS from the central server. A service provider or an enterprise with large deployment of HT502 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 Networks, Inc. HT-502 User Manual Firmware Version 1.0.1.21 Page 27 of 31 Last Updated: 12/2008
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.
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SOFTWARE UPGRADE

Software upgrade can be done via either TFTP or HTTP. The corresponding configuration settings are in the ADVANCED SETTINGS configuration page.
Firmware Upgrade through TFTP/HTTP/HTTPS
To upgrade via TFTP or HTTP/HTTPS, the Firmware Upgrade and Provisioning upgrade via field needs to be set to TFTP HTTP or HTTPS, respectively. Firmware Server Path needs to be set to a valid URL of a TFTP or 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/1.0.1.21 e.g. 168.75.215.189 NOTES: Firmware upgrade server in IP address format can be configured via IVR. Please refer to the CONFIGURATION GUIDE section for instructions. If the server is in FQDN format, it must be set via the web configuration interface. Grandstream recommends end-user use the Grandstream TFTP server. Its address can be found at http://www.grandstream.com/firmware.html. Currently the TFTP firmware server IP address is 168.75.215.189. For large companies, we recommend to maintain their own TFTP/ HTTP/HTTPS server for upgrade and provisioning procedures. Once a Firmware Server Path is set, user needs to update the settings and reboot the device. If the configured firmware server is found and a new code image is available, the HT502 will attempt to retrieve the new image files by downloading them into the GXW400x s SRAM. During this stage, the HT502s LEDs will blink until the checking/downloading process is completed. Upon verification of checksum, the new code image will then be saved into the Flash. If TFTP/HTTP/HTTPS fails for any reason (e.g. TFTP/HTTP/HTTPS server is not responding, there are no code image files available for upgrade, or checksum test fails, etc), the HT502 will stop the TFTP/HTTP/HTTPS process and simply boot using the existing code image in the flash. Firmware upgrade may take as long as 15 to 30 minutes over Internet, or just 5 minutes 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. Grandstreams latest firmware is available http://www.grandstream.com/firmware.html. Oversea users are strongly recommended to download the binary files and upgrade firmware locally in a controlled LAN environment. 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.

Managing Firmware and Configuration File Download
When Automatic Upgrade is set Yes, every the auto check will be done in the minute specified in this field. If set to daily at hour (0-23), Service Provider can use P193 (Auto Check Interval) to have the devices do a daily check at the hour set in this field with either Firmware Server or Config Server. If set to weekly on day (0-6) the auto check will be done in the day specified in this field. This allows the device periodically check if there are any new changes need to be taken on a scheduled time. By defining different intervals in P193 for different devices, Server Provider can spread the Firmware or Configuration File download in minutes to reduce the Firmware or Provisioning Server load at any given time.
Automatic Upgrade: No Yes, every

1 10080

minutes(60-5256000). Yes, weekly on day

Yes, daily at hour

(0-23).

(0-6).

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RESTORE FACTORY DEFAULT SETTING
WARNING! Restoring the Factory Default Setting will DELETE all configuration information of the phone. Please BACKUP or PRINT out all the settings before you approach to following steps. Grandstream will not take any responsibility if you lose all the parameters of setting and cannot connect to your VoIP service provider. FACTORY RESET There are two (2) methods for resetting your unit: Reset Button Reset default factory settings following these four (4) steps: 1. Unplug the Ethernet cable. 2. Locate a needle-sized hole on the back panel of the gateway unit next to the power connection. 3. Insert a pin in this hole, and press for about 7 seconds. 4. Take out the pin. All unit settings are restored to factory settings.
IVR Command Reset default factory settings using the IVR Prompt (Table 5): 1. 2. 3. 4. Dial *** for voice prompt. Enter 99 and wait for reset voice prompt. Enter the encoded MAC address (Look below on how to encode MAC address). Wait 15 seconds and device will automatically reboot and restore factory settings.

doc1

Grandstream Networks, Inc.
HT502 Dual FXS Port Analog Telephone Adaptor
HT502 User Manual Firmware Version 1.0.1.63
www.grandstream.com support@grandstream.com
TABLE OF CONTENTS HT502 User Manual WELCOME..... 4 SAFETY COMPLIANCES.... 4 WARRANTY..... 4 CONNECT YOUR HT502..... 5 EQUIPMENT PACKAGING.... 5 CONNECTING THE HT502..... 5 PRODUCT OVERVIEW..... 7 SOFTWARE FEATURES OVERVIEW.... 7 HARDWARE SPECIFICATION.... 8 BASIC OPERATIONS..... 9 UNDERSTANDING HT502 VOICE PROMPT.... 9 PLACING A PHONE CALL..... 10
Phone or Extension Numbers...10 Direct IP Calls....10
CALL HOLD.... 11 CALL WAITING.... 11 CALL TRANSFER..... 11 3-WAY CONFERENCING..... 12 FAX SUPPORT.... 12 CALL FEATURES.... 13 CONFIGURATION GUIDE.... 14 CONFIGURING THE HT502 THROUGH VOICE PROMPTS... 14 CONFIGURING THE HT502 VIA WEB BROWSER... 15
Access the Web Configuration Menu...15
IMPORTANT SETTINGS..... 16
NAT Settings....16 DTMF Methods....16 Set priority of DTMF methods according to your preference. This setting should be based on your server DTMF setting....16 Preferred VOCODER (Codec)...16
ADVANCED USER CONFIGURATION.... 20 SAVING THE CONFIGURATION CHANGES.... 27 REBOOTING THE HT502 FROM REMOTE.... 27 CONFIGURATION THROUGH A CENTRAL SERVER... 28 SOFTWARE UPGRADE..... 29 FIRMWARE UPGRADE THROUGH TFTP/HTTP/HTTPS... 29 CONFIGURATION FILE DOWNLOAD.... 30 FIRMWARE AND CONFIGURATION FILE PREFIX AND POSTFIX.. 30 MANAGING FIRMWARE AND CONFIGURATION FILE DOWNLOAD... 30 RESTORE FACTORY DEFAULT SETTING.... 31
HT-502 User Manual Firmware Version 1.0.1.63
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TABLE OF FIGURES HT502 USER MANUAL FIGURE 1: CONNECTING THE HT502.... 5 FIGURE 2: HT502 CONNECTION DIAGRAM..... 6 FIGURE 3: UPLINK/DOWNLINK BANDWIDTH LIMITATION.19
TABLE OF TABLES HT502 USER MANUAL TABLE 1: DEFINITIONS OF THE HT502 CONNECTORS... 6 TABLE 2: DEFINITIONS OF THE HT502 LEDS.... 6 TABLE 3: HT502 TECHNICAL SPECIFICATIONS.... 7 TABLE 5: HT502 IVR MENU DEFINITIONS.... 9 TABLE 6: HT502 CALL FEATURES.... 13 TABLE 7: BASIC SETTINGS.... 17 TABLE 8: STATUS PAGE..... 19 TABLE 9: ADVANCED SETTINGS..... 20 TABLE 10: ACCOUNT SETTINGS..... 22
CONFIGURATION GUI INTERFACE EXAMPLES
HT502 USER MANUAL (http://www.grandstream.com/support/ht_series/ht502/documents/ht502_gui.zip) 1. SCREENSHOT OF ADVANCED USER CONFIGURATION PAGE 2. SCREENSHOT OF BASIC SETTINGS CONFIGURATION PAGE 3. SCREENSHOT OF FXS PORT 1 CONFIGURATION LOGIN PAGE 4. SCREENSHOT OF FXS PORT 2 CONFIGURATION PAGE 5. SCREENSHOT OF STATUS PAGE 6. 7.
SCREENSHOT OF LOGIN PAGE SCREENSHOT OF REBOOT PAGE
8. SCREENSHOT OF REBOOTING PAGE
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WELCOME

Thank you for purchasing Grandstreams HT502, the affordable, feature rich Analog Telephone Adaptor. Grandstream HandyTone502 is a new addition to the popular HandyTone ATA product family. It features the rich audio quality, a broad range of voice codecs, and functionality of the HT502, including two (2) FXS ports each with independent SIP accounts. This manual will help you learn how to operate and manage your HandyTone502 Analog Telephone Adaptor and make the best use of its many upgraded features including simple and quick installation, 3way conferencing, and direct IP-IP Calling. This HT502 is very easy to manage and configure, and is specifically designed to be an easy to use and affordable VoIP solution for both the residential user and the teleworker.

Safety Compliances

The HT502 phone complies with FCC/CE and various safety standards. The HT502 power adaptor is compliant with UL standard. Only use the universal power adapter provided with the HT502 package. The manufacturers warranty does not cover damages to the phone caused by unsupported power adaptors.

Warranty

If you purchased your HT502 from a reseller, please contact the company where you purchased your device for replacement, repair or refund. If you purchased the product directly from Grandstream, contact your Grandstream Sales and Service Representative for a RMA (Return Materials Authorization) number before you return the product. Grandstream reserves the right to remedy warranty policy without prior notification. Caution: Changes or modifications to this product not expressly approved by Grandstream, or operation of this product in any way other than as detailed by this User Manual, could void your manufacturer warranty. Please do not use a different power adaptor with the HT502 as it may cause damage to the products and void the manufacturer warranty.

This document contains links to HT502 GUI Interfaces. Please download these examples from http://www.grandstream.com/support/ht_series/ht502/documents/ht502_gui.zip for your reference. This document is subject to change without notice. The latest electronic version of this user manual is available for download at: http://www.grandstream.com/support/ht_series/ht502/documents/ht502_usermanual_english.pdf
Reproduction or transmittal of the entire or any part, in any form or by any means, electronic or print, for any purpose is not permitted without the express written permission of Grandstream Networks, Inc.
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CONNECT YOUR HT502

Equipment Packaging
The HT502 ATA package contains: One HT502 Main Case One Universal Power Adaptor One Ethernet Cable One HT502 Vertical Stand

Connecting the HT502

The HT502 is designed for easy configuration and easy installation. Configure the HT502 following the directions in the Configuration section of this manual. 1. Connect a standard touch-tone analog telephone to the PHONE port. 2. Insert a standard RJ11 telephone cable into the Phone1 port and connect the other end of the telephone cable to the analog telephone. 3. Insert the Ethernet cable into the WAN port of HT502 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 HT502 if it is being used as a router. 5. Insert the power adapter into the HT502 and connect it to a wall outlet. The HT502 Analog Telephone Adaptor is an all-in-one VoIP integrated device designed to be a total solution for networks providing VoIP services. The HT502 VoIP features and functions are available using a regular analog telephone. Figure 1: Connecting the HT502

HT-502 Front View

HT-502 Back View

Display LEDs (green)

RJ-45 Ports 10/100 Mbps

Power Supply (12V)

RJ-11 FXS Ports (Phone)
The HT502 has two FXS port. Both FXS ports can have a separate SIP account. This is a key feature of HT502 as it supports simultaneous calls on both FXS ports.
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Voice Compression

DHCP Server/Client Telnet Server Fax over IP QoS IP Transport
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DTMF Method IP Signaling Provisioning Control Device Management
Flexible DTMF transmission method, user interface of In-audio, RFC2833, and/or SIP Info SIP (RFC 3261) TFTP, HTTP, HTTPS TLS/SIPS , SIP over TCP/TLS Web interface or via secure encrypted AES or non-encrypted central configuration file for mass deployment, Auto/manual provisioning system or via built-in IVR. NAT-friendly remote software upgrade (via TFTP/HTTP/HTTPS) for deployed devices including behind firewall/NAT. Syslog support Yes Yes Input: 100240 VAC/50-60 Hz 0.3A Max Output: 12VDC, 0.5A, UL certified Operational: 32o104oF or 0o40oC Storage: 10o130o F / Humidity: 1090% Non-condensing
Dial Plan UPnP Universal Switching Power Adaptor Environmental
Dimensions (H x W x D) Short Haul Loop Call Handling Features
115mm (L) x 75mm (W) x 27mm (H) 2REN, Up to 1Km on 24 AWG wire Caller ID display or block, Call waiting caller ID, Call waiting/flash, Call transfer, hold, forward, mute, 3-way conferencing, message waiting, Do-Not-Disturb (DND), call-return service Bellcore Type 1 & 2, ETSI, BT, NTT, and DTMF-based CID Yes EN55022/EN55024 and FCC part15 Class B UL
Caller ID Polarity Reversal / Wink EMC Safety

Hardware Specification

The table below lists the hardware specification of HT502. TABLE 4: HT502 HARDWARE SPECIFICATION
LAN Interface LED Universal Switching Power Adaptor 2 x RJ45 10/100Mbps (integrated router) 5 LEDs (GREEN) Input: 100-240V AC, 50/60Hz, 0.5A Max Output: 12V DC, 1.25A UL certified Dimension Weight Temperature Humidity Compliance 115mm (L) x 75mm (W) x 27mm (H) 94 g (0.21lbs) 32~104F / 0~40C 10% - 90% (non-condensing)
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BASIC OPERATIONS

Understanding HT502 Voice Prompt
HT502 has a built-in voice prompt menu for simple device configuration. The IVR menu and the LED button work with any of the FXS port. Pick up the handset and dial *** to use the IVR menu. TABLE 5: HT502 IVR MENU DEFINITIONS
Menu Main Menu Voice Prompt Enter a Menu Option Options Press * for the next menu option Press # to return to the main menu Enter 01-05, 07,10, 12-17,47 or 99 menu options Press 9 to toggle the selection If using Static IP Mode, configure the IP address information using menus 02 to 05. If using Dynamic IP Mode, all IP address information comes from the DHCP server automatically after reboot. The current WAN IP address is announced If using Static IP Mode, enter 12 digit new IP address. You need to reset the HT to take affect the new IP address. Same as menu 02 Same as menu 02 Same as menu 02 Press 9 to move to the next selection in the list: PCM U / PCM A iLBC G-726 G-723 G-729 Announces the Mac address of the unit. Press 9 to toggle between enable / disable Announces current Firmware Server IP address. Enter 12 digit new IP address. Announces current Config Server Path IP address. Enter 12 digit new IP address. Upgrade protocol for firmware and configuration update. Press 9 to toggle between TFTP / HTTP / HTTPS Firmware version information. Firmware upgrade mode. Press 9 to toggle among the following three options: - always check - check when pre/suffix changes - never upgrade Enter the target IP address to make a direct IP call, after dial tone. (See Make a Direct IP Call.) User can make internal calls between two FXS ports on the same HT502 even without being registered to SIP server. By dialing *** and 70 user can reach the other FXS port Number of Voice Mails HT-502 User Manual Firmware Version 1.0.1.63 Page 9 of 31 Last Updated: 11/2010

DHCP Mode, Static IP Mode

IP Address + IP address

Subnet + IP address Gateway + IP address DNS Server + IP address Preferred Vocoder
MAC Address WAN Port Web Access Firmware Server IP Address Configuration Server IP Address Upgrade Protocol Firmware Version Firmware Upgrade
Direct IP Calling Phone calls between FXS 1 and FXS 2 port

Voice Mail

Press 9 to reboot the device Enter MAC address to restore factory default setting (See Restore Factory Default Setting section) Automatically returns to main menu This prompt will be played immediately after off hook If the device is not register and the option Outgoing Call without Registration is in NO
Invalid Entry Device not registered
Five Success Tips when using the Voice Prompt 1. * shifts down to the next menu option 2. # returns to the main menu 3. 9 functions as the ENTER key in many cases to confirm an option 4. All entered digit sequences have known lengths - 2 digits for menu option and 12 digits for IP address. For IP address, add 0 before the digits if the digits are less than 3 (i.e. - 192.168.0.26 should be key in like 192168000026. No decimal is needed). 5. Key entry can not be deleted but the phone may prompt error once it is detected

Placing a Phone Call

PHONE OR EXTENSION NUMBERS 1. Dial the number directly and wait for 4 seconds (Default No Key Entry Timeout); or 2. Dial the number directly and press # (Use # as dial key must be configured in web configuration). Examples: 1. Dial an extension directly on the same proxy, (e.g. 1008), and then press the # or wait for 4 seconds. 2. Dial an outside number (e.g. (626) 666-7890), first enter the prefix number (usually 1+ or international code) followed by the phone number. Press # or wait for 4 seconds. Check with your VoIP service provider for further details on prefix numbers.
DIRECT IP CALLS Direct IP calling allows two parties, that is, a FXS Port with an analog phone and another VoIP Device, to talk to each other in an ad hoc fashion without a SIP proxy. Elements necessary to completing a Direct IP Call: 1. Both HT502 and other VoIP Device, have public IP addresses, or 2. Both HT502 and other VoIP Device are on the same LAN using private IP addresses, or 3. Both HT502 and other VoIP Device can be connected through a router using public or private IP addresses (with necessary port forwarding or DMZ). HT502 supports two ways to make Direct IP Calling: Using IVR 1. Pick up the analog phone then access the voice menu prompt by dial *** 2. Dial 47 to access the direct IP call menu 3. Enter the IP address after the dial tone and voice prompt Direct IP Calling

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Using Star Code 1. Pick up the analog phone then dial *47 2. Enter the target IP address. Note: NO dial tone will be played between step 1 and 2. Destination ports can be specified using * (encoding for :) followed by the port number. Examples of Direct IP Calls: a) If the target IP address is 192.168.0.160, the dialing convention is *47 or Voice Prompt with option 47, then 192*168*0*160. followed by pressing the # key if it is configured as a send key or wait 4 seconds. In this case, the default destination port 5060 is used if no port is specified. b) If the target IP address/port is 192.168.1.20:5062, then the dialing convention would be: *47 or Voice Prompt with option 47, then 192*168*0*160*5062 followed by pressing the # key if it is configured as a send key or wait for 4 seconds. NOTE: When completing direct IP call, the Use Random Port should set to NO. You cannot make direct IP calls between FXS1 to FXS2 since they are using same IP.

Call Hold

Place a call on hold by pressing the flash button on the analog phone (if the phone has that button). Press the flash button again to release the previously held Caller and resume conversation. If no flash button is available, use hook flash (toggle on-off hook quickly). You may drop a call using hook flash.

Call Waiting

Call waiting tone (3 short beeps) indicates an incoming call, if the call waiting feature is enabled. Toggle between incoming call and current call by pressing the flash button. First call is placed on hold. Press the flash button to toggle between two active calls.

Call Transfer

Blind Transfer Assume that call Caller A and B are in conversation. A wants to Blind Transfer B to C: 3. Caller A presses FLASH on the analog phone to hear the dial tone. 4. Caller A dials *87 then dials caller Cs number, and then # (or wait for 4 seconds) 5. Caller A will hear the confirm tone. Then, A can hang up. NOTE: Enable Call Feature must be set to Yes in web configuration page. Caller A can place a call on hold and wait for one of three situations: 1. A quick confirmation tone (similar to call waiting tone) followed by a dial-tone. This indicates the transfer is successful (transferee has received a 200 OK from transfer target). At this point, Caller A can either hang up or make another call. 2. A quick busy tone followed by a restored call (on supported platforms only). This means the transferee has received a 4xx response for the INVITE and we will try to recover the call. The busy tone is just to indicate to the transferor that the transfer has failed. 3. Continuous busy tone. The phone has timed out.
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Note: continuous busy tone does not indicate the transfer has been successful, nor does it indicate the transfer has failed. It often means there was a failure to receive second NOTIFY check firmware for most recent release. Attended Transfer Assume that Caller A and B are in conversation. Caller A wants to Attend Transfer B to C: 1. Caller A presses FLASH on the analog phone for dial tone. 2. Caller A then dials Caller Cs number followed by # (or wait for 4 seconds). 3. If Caller C answers the call, Caller A and Caller C are in conversation. Then A can hang up to complete transfer. 4. If Caller C does not answer the call, Caller A can press flash to resume call with Caller B. NOTE: When Attended Transfer fails and A hangs up, the HT502 will ring back user A to remind A that B is still on the call. A can pick up the phone to resume conversation with B.

Web Port Telnet Server IP Address

DHCP hostname

DHCP domain
DHCP vendor class ID PPPoE account ID
PPPoE password PPPoE Service Name
Time Zone Self Defined Time Zone

Device Mode

NAT Maximum Ports

NAT TCP Timeout

NAT TCP idle timeout in seconds. Connection will be closed after preconfigured, timeout if not refreshed. Range: 0 - 3600 NAT TCP idle timeout in seconds. Connection will be closed after preconfigured, timeout if not refreshed. Range: 0 3600, default is 300 The maximum uplink bandwidth permitted by the device. This function is disabled by default. The total bandwidth can be set as: 128K, 256K, 512K, 1M, 2M, 3M, 4M, 5M, 10M or 15M. The primary function of this setting is to limit the uplink bandwidth for the device internal system, signaling and NATed traffic. Example: if 512k is configured, there will be at least 512kbps limited for internal system, signaling and NATed traffic. Voice or RTP stream will never be limited. See figure 3. The maximum downlink bandwidth permitted by the device. This function is disabled by default. The total bandwidth can be set as: 128K, 256K, 512K, 1M, 2M, 3M, 4M, 5M, 10M or 15M. The primary function of this setting is to limit the download bandwidth for the device internal system, signaling and NATed traffic. Example: if 128 is configured, there will be at least 128kbps limited for internal system, signaling and NATed traffic. Voice or RTP stream will never be limited. See figure 3. When set to Yes, the HT502 acts as an UPnP gateway for your UPnP enabled applications. UPnP = Universal Plug and Play Default is No. When set to Yes, the HT502 responds to the PING command from other computers, but is also made vulnerable to DOS attacks. Default is No. When set to Yes, the user can access the web configuration pages through the WAN port, instead of through the PC port. Warning: this configuration is less secure than the default option. This allows the user to change/set a specific MAC address on the WAN interface. Note: Set in Hex format When set to Yes, device will function as a simple router and LAN port will provide IP addresses to internal network. Connect the WAN port to ADSL/Cable modem or any other equipment that provides access to public Internet. Base IP Address for a LAN port. Default factory setting is 192.168.2.1 Default value is 100. The last segment of IP address assigned to the HT50x in the LAN Network. Default configuration assigns IP address (to local network devices) starting from 192.168.2.100. Default value is 199. This parameter allows a user to limit the number of local network devices connected to the internal router. Default configuration assigns IP address (to devices connected to the LAN port) in a range from 192.168.2.100 up to 192.168.2.199. Sets the LAN subnet mask. Default value is 255.255.255.0 Base IP address for the LAN port, which functions as default gateway for its LAN. Default value is 192.168.2.1 Default value is 120 hrs (5 days). The length of time the IP address is assigned to the LAN clients. Value is set in units of hours. This function forwards all WAN IP traffic to a specific IP address if no matching port is used by HT502 or in the defined port forwarding. Forwards a matching (TCP/UDP) port to a specific LAN IP address with a specific (TCP/UDP) port.

SIP transport NAT Traversal (STUN)
Authenticate Password Name DNS Mode
SIP service subscribers account password. SIP service subscribers name for Caller ID display. One from the 3 modes are available for DNS Mode configuration: -A Record (for resolving IP Address of target according to domain name) -SRV (DNS SRV resource records indicates how to find services for various protocols) -NAPTR/SRV (Naming Authority Pointer according to RFC 2915) One mode can be chosen for the client to look up server. The default value is A Record
User ID is Phone Number SIP Registration Unregister on Reboot Outgoing Call without Registration Register Expiration
If the HT502 has an assigned PSTN telephone number, this field should be set to Yes. Otherwise, set to No. If set to Yes, a user=phone parameter will be appended to the From header in SIP request. Controls whether the HT502 needs to send REGISTER messages to the proxy server. The default setting is Yes. Default is No. If set to Yes, the SIP users registration information will be cleared on reboot. Default is No. If set to Yes, user can place outgoing calls even when not registered (if allowed by Internet Telephone Service Provider) but is unable to receive incoming calls. This parameter allows the user to specify the time frequency (in minutes) the HT502 refreshes its registration with the specified registrar. The default interval is 60 minutes (or 1 hour). The maximum interval is 65535 minutes (about 45 days). Retry registration if the process failed. Default is 30 seconds. Defines the local SIP port the HT502 will listen and transmit. The default value for FXS port 1 is 5060. The default value for FXS port 2 is 5062. Defines the local RTP-RTCP port pair the HT502 will listen and transmit. It is the base RTP port for channel 0. When configured, channel 0 uses this port _value for RTP and the port_value+1 for its RTCP; channel 1 uses port_value+2 for RTP and port_value+3 for its RTCP. The default value for FXS port 1 is 5004. The default value for FXS port 2 is 5012.
Registration Retry Wait Time Local SIP port Local RTP port
Use Random Port Refer to Use Target Contact Transfer on Conference Hang up
This parameter forces the random generation of both the local SIP and RTP ports when set to Yes. This is usually necessary when multiple HT502 are behind the same NAT. Default is NO. If set to YES, then for Attended Transfer, the Refer-To header uses the transferred targets Contact header information. Default is No. In which case if the conference originator hangs up the conference will be terminated. When option YES is chosen, originator will transfer other parties to
each other so that B and C can choose either to continue the conversation or hang up.
Remove OBP from Route Header Support SIP Instance ID Validate incoming SIP message Default is No. When option YES is chosen, the Out Bound Proxy will be removed from Route header. Default is Yes. If set to Yes, the contact header in REGISTER request will contain SIP Instance ID as defined in IETF SIP Outbound draft. Default is No. If set to yes all incoming SIP messages will be strictly validated according to RFC rules. If message will not pass validation process, call will be rejected. Default is No. Check the incoming SIP User ID in Request URI. If they dont match, the call will be rejected. If this option is enabled, the device will not be able to make direct IP calls. T1 is an estimate of the round-trip time between the client and server transactions. HT-502 User Manual Firmware Version 1.0.1.63 Page 23 of 31 Last Updated: 11/2010

Disable Call Waiting Disable Call-Waiting Caller ID Disable Call Waiting Tone Disable Reminder Ring for On-Hold Call Disable Visual MWI Ring Timeout Delayed Call Forward Wait Time No Key Entry Timeout Early Dial
Default is No. If set to YES Call Waiting indication information will not be provided to analog phone connected to this FXS port. Default is No. If set to YES Call Waiting caller ID will not be provided to analog phone connected to this FXS port. Default is No. This is to disable the stutter Call Waiting Tone when a Call Waiting information arrives. The CWCID information will still be displayed. Default is No. Do not play the reminder ring when this is set to Yes. If set to Yes, the MWI information will not be transferred to the analog phone connected to the FXS port. Incoming call will stop ringing when not picked up given a specific period of time. Default value is 20 seconds. In case this feature activated using * codes (*92 code), the call will be forwarded after this preconfigured amount of time. Default is 4 seconds. Dialing process is completed and outgoing call is initiated if no key entry occurs during this preconfigured interval. Default is No. Use only if proxy supports 484 response. This parameter controls whether the phone will send an early INVITE each time a key is pressed when a user dials a number. If set to Yes, an INVITE is sent using the dial-number collected thus far; Otherwise, no INVITE is sent until the (Re-)Dial button is pressed or after about 5 seconds have elapsed if the user forgets to press the Re-Dial button. The Yes option should be used ONLY if there is a SIP proxy configured and the proxy
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server supports 484 Incomplete Address response. Otherwise, the call will likely be rejected by the proxy (with a 404 Not Found error). This feature does NOT work with and should NOT be enabled for direct IP-to-IP calling. Sets the prefix added to each dialed number. Allows users to configure the # key as the Send (or Dial) key. If set to Yes, # will send the number. In this case, this key is essentially equivalent to the Dial key. If set to No, this # key can be included as part of number.

Dial Plan Prefix Use # as Dial Key

Dial Plan

Dial Plan Rules: 1. Accept Digits: 1,2,3,4,5,6,7,8,9,0 , *, #, A,a,B,b,C,c,D,d 2. Grammar: x - any digit from 0-9; xx+ - at least 2 digits number; xx. at least 1 digit number. ^ - exclude; [3-5] - any digit of 3, 4, or 5; [147] - any digit 1, 4, or 7; <2=011> - replace digit 2 with 011 when dialing | - or Example 1: {[369]11 | 1617xxxxxxx} Allow 311, 611, 911, and any 11 digit numbers with leading digits 1617 Example 2: {^1900x+ | <=1617>xxxxxxx} Block any number of leading digits 1900 and add prefix 1617 for any dialed 7 digit numbers Example 3: {1xxx[2-9]xxxxxx | <2=011>x+} Allow any combinations of numbers with 11 digits which has a leading digit 1, but 5th digit cannot be 0 or 1. Or any length of numbers with a minimum of 2 digits beginning with 2, with the leading digit replaced with 011. 3. Default: Outgoing - {x+} Example of a simple dial plan used in a Home/Office in the US: { ^1900x. | <=1617>[2-9]xxxxxx | 1[2-9]xx[2-9]xxxxxx | 011[2-9]x. | [3469]11 } Explanation of example rule (reading from left to right): ^1900x. - prevents dialing any number started with 1900 <=1617>[2-9]xxxxxx - allows dialing to local area code (617) numbers by dialing 7 numbers and 1617 area code will be added automatically 1[2-9]xx[2-9]xxxxxx |- allows dialing to any US/Canada Number with 11 digits length 011[2-9]x. - allows international calls starting with 011 [3469]11 - allow dialing special and emergency numbers 311, 411, 611 and 911 Note: In some cases user wishes to dial strings such as *123 to activate voice mail or other application provided by service provider. In this case * should be predefined inside dial plan feature and the Dial Plan should be: { *x+ }.
Subscribe for MWI Send Anonymous
Default is No. When set to Yes a SUBSCRIBE for Message Waiting Indication will be sent periodically. If this parameter is set to Yes, the From header along with Privacy and P_ Asserted_Identity headers in outgoing INVITE message will be set to anonymous, blocking Caller ID. Default is No. If set to Yes, incoming calls with anonymous Caller ID will be rejected with 486 Busy message. Default is Standard. Choose the selection to meet some special requirements from Softswitch vendors.
Anonymous Call Rejection Special Feature

Instructions for local firmware upgrade: 1. Unzip the file and put all of them under the root directory of the TFTP server. 2. Put the PC running the TFTP server and the HT502 device in the same LAN segment.
Grandstream Networks, Inc. HT-502 User Manual Firmware Version 1.0.1.63 Page 29 of 31 Last Updated: 11/2010
3. Please go to File -> Configure -> Security to change the TFTP server's default setting from "Receive Only" to "Transmit Only" for the firmware upgrade. 4. Start the TFTP server, in the phones web configuration page 5. Configure the Firmware Server Path with the IP address of the PC 6. Update the change and reboot the unit End users can also choose to download the free HTTP server from http://httpd.apache.org/ or use Microsoft IIS web server.
Configuration File Download
Grandstream SIP Device can be configured via Web Interface as well as via Configuration File through TFTP or HTTP/HTTPS. Config Server Path is the TFTP or HTTP/HTTPS server path for configuration file. It needs to be set to a valid URL, either in FQDN or IP address format. The Config Server Path can be same or different from the Firmware Server Path. A configuration parameter is associated with each particular field in the web configuration page. A parameter consists of a Capital letter P and 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. For a detailed parameter list, please refer to the corresponding firmware release configuration template. When Grandstream Device boots up or reboots, it will issue request for configuration file named cfgxxxxxxxxxxxx, where xxxxxxxxxxxx is the LAN MAC address of the device, i.e., cfg000b820102ab. The configuration file name should be in lower cases.
Firmware and Configuration File Prefix and Postfix
Firmware Prefix and Postfix 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.

Managing Firmware and Configuration File Download
When Automatic Upgrade is set Yes, every the auto check will be done in the minute specified in this field. If set to daily at hour (0-23), Service Provider can use P193 (Auto Check Interval) to have the devices do a daily check at the hour set in this field with either Firmware Server or Config Server. If set to weekly on day (0-6) the auto check will be done in the day specified in this field. This allows the device periodically check if there are any new changes need to be taken on a scheduled time. By defining different intervals in P193 for different devices, Server Provider can spread the Firmware or Configuration File download in minutes to reduce the Firmware or Provisioning Server load at any given time.
Automatic Upgrade: No Yes, every

1 10080

minutes(60-5256000). Yes, weekly on day

Yes, daily at hour

(0-23).

(0-6).

Page 30 of 31 Last Updated: 11/2010
RESTORE FACTORY DEFAULT SETTING
WARNING! Restoring the Factory Default Setting will DELETE all configuration information of the phone. Please BACKUP or PRINT out all the settings before you approach to following steps. Grandstream will not take any responsibility if you lose all the parameters of setting and cannot connect to your VoIP service provider. FACTORY RESET There are two (2) methods for resetting your unit: Reset Button Reset default factory settings following these four (4) steps: 1. Unplug the Ethernet cable. 2. Locate a needle-sized hole on the back panel of the gateway unit next to the power connection. 3. Insert a pin in this hole, and press for about 7 seconds. 4. Take out the pin. All unit settings are restored to factory settings.
IVR Command Reset default factory settings using the IVR Prompt (Table 5): 1. 2. 3. 4. Dial *** for voice prompt. Enter 99 and wait for reset voice prompt. Enter the encoded MAC address (Look below on how to encode MAC address). Wait 15 seconds and device will automatically reboot and restore factory settings.
Encode the MAC Address 1. Locate the MAC address of the device. It is the 12 digit HEX number on the bottom of the unit. 2. Key in the MAC address. Use the following mapping: 0-9: 0-9 A: 22 (press the 2 key twice, A will show on the LCD) B: 222 C: 2222 D: 33 (press the 3 key twice, D will show on the LCD) E: 333 F: 3333 For example: if the MAC address is 000b8200e395, it should be keyed in as 0002228200333395. NOTE: 1. Factory Reset will be disabled if the Lock keypad update is set to Yes. 2. Please be aware by default the HT502 WAN side HTTP access is disabled. After a factory reset, the devices web configuration page can be accessed only from its LAN port.

3. If the HT502 was previously locked by your local service provider, pressing the RESET button will only
restart the unit. The device will not return to factory default settings.
Page 31 of 31 Last Updated: 11/2010

 

Technical specifications

General
Device TypeVoIP phone adapter
Networking
Form FactorExternal
Ports Qty2
Connectivity TechnologyWired
Data Transfer Rate100 Mbps
Data Link ProtocolEthernet, Fast Ethernet
Remote Management ProtocolTelnet, HTTP, HTTPS
Status IndicatorsPort status, power, phone
FeaturesSwitching, routing, DHCP support, NAT support, VLAN support, DiffServ support
Compliant StandardsIEEE 802.1Q, IEEE 802.1p
IP Telephony
VoIP ProtocolsSIP
Voice CodecsG.711, G.723.1, G.728, G.729a, G.729ab, G.711u, G.711a, G.726, iLBC, T.38
Telephony Interfaces2 phone (FXS)
IP Telephony FeaturesEcho cancellation (G.168), caller ID, call waiting
Expansion / Connectivity
Interfaces1 x network - Ethernet 10Base-T/100Base-TX - RJ-45 1 x network - Ethernet 10Base-T/100Base-TX - RJ-45 ( WAN ) 2 x phone line - FXS - RJ-11
Miscellaneous
Cables Included1 x network cable
Compliant StandardsUL, EN55022, EN55024, FCC Part 15 B
Power
Power DevicePower adapter - external
Voltage RequiredAC 120/230 V ( 50/60 Hz )
Environmental Parameters
Min Operating Temperature32 °F
Max Operating Temperature104 °F
Humidity Range Operating10 - 90%
Universal Product Identifiers
BrandGrandstream Networks
Part NumberHT-502

 

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