Grandstream Networks, Inc. HT503 FXS/FXO Port Analog Telephone Adaptor HT503 User Manual Firmware Version 1.0.0.6 www.grandstream.com support@grandstream.
TABLE OF CONTENTS HT503 USER MANUAL WELCOME....................................................................................................................................................... 4 Safety Compliances.................................................................................................................................. 4 Warranty ...................................................................................................................................................
TABLE OF FIGURES HT503 USER MANUAL FIGURE 1: FIGURE 2: FIGURE 3: FIGURE 4: CONNECTING THE HT503 ............................................................................................................... 5 INTERCONNECTION DIAGRAM OF THE HT503 ................................................................................... 6 SCREENSHOT OF CONFIGURATION LOG-IN PAGE ............................................................................ 19 SCREENSHOT OF REBOOTING SCREEN............................
WELCOME Thank you for purchasing Grandstream’s HT503, the affordable, feature rich, Analog Telephone Adaptor/IAD. The HT503 combines a sleek design with the latest technology to offer more advanced telephony features and significantly better integrated router performance than its predecessor – the HT488. It is the second ATA/IAD in the HandyTone 50x series. The HT503 functions as a true 3-in-1 gateway for PSTN network, analog telephone FXS interface and IP network.
INSTALLATION EQUIPMENT PACKAGING The HT503 ATA package contains: • • • • One HT503 Main Case One Universal Power Adaptor One Ethernet Cable One HT503 Vertical Stand CONNECTING YOUR ATA The HT503 Analog Telephone Adaptor is an all-in-one VoIP integrated device designed to be a total solution for networks providing VoIP services. The HT503 VoIP features and functions are available using a regular analog telephone.
FIVE EASY STEPS TO INSTALL THE HT503 The HT503 is designed for easy configuration and easy installation. Configure the HT503 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 LINE port and connect the other end of the telephone cable to a wall jack. 3.
PRODUCT OVERVIEW The HT503 is an affordable, high-quality, integrated IP telephony solution for both the residential customers and the ‘road-warriors’ who need advanced call features between traditional PSTN network and IP network. The HT503 enables IP connectivity for any phone or fax using the FXS port and a webbased GUI for easy configuration and installation.
HARDWARE SPECIFICATION The table below lists the hardware specification of HT503. TABLE 3: HT503 HARDWARE SPECIFICATION LAN interface 1xRJ45 10/100 Mbps Port WAN interface 1xRJ45 10/100 Mbps Port FXS telephone port 1 x FXS (RJ11) FXO telephone port (PSTN Port) 1x PSTN pass-through and life line port LED Power, WAN, LAN, PHONE, and LINE (Green) Universal Switching Power Adaptor Input: 100–240 VAC, 50-60 Hz Output: 12VDC, 0.
BASIC OPERATIONS GET FAMILIAR WITH VOICE PROMPT HT503 has a stored voice prompt menu for quick browsing and simple configuration. Currently, the voice prompt menu is designed for the FXS port only. Dial “***” from the analog phone to enter the voice prompt.
NOTE: • • • • • “*” shifts down to the next menu option “#” returns to the main menu “9” functions as the ENTER key in many cases to confirm an option All entered digit sequences have known lengths - 2 digits for menu option and 12 digits for IP address. For IP address, add 0 before the digits if the digits are less than 3 (like 192.168.0.26 should be key in like 192168000026, no dot needed while input). Once all of the digits are collected, the input will be processed.
EXAMPLES: 1. If the target IP address is 192.168.0.10, the dialing convention is Voice Prompt with option 47, then 192 168 000 010 followed by pressing the “#” key if it is configured as a send key or wait for more than 5 seconds. 2. 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.
Three situations can follow the transfer: 1. A quick confirmation tone (temporarily using the call waiting indication tone) followed by a dialtone. This indicates the transfer was successful (transferee has received a 200 OK from transfer target). 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.
1. A presses FLASH (on the analog phone, or Hook Flash for old model phones) to get a dial tone. 2. A dials C’s number then # (or wait for 4 seconds). 3. If C answers the call, then A presses FLASH to bring B, C in the conference. 4. If C does not answer the call, A can press FLASH back to talk to B. 5. If A presses FLASH during the conference, C will be dropped out. Note: Party A is the call initiator for both calls with party B and party C.
PSTN-TO-VOIP CALLS This function is available using the FXO port. The FXO port functions as a bridge between the Internet and PSTN and enables calls to be passed from the PSTN network to VoIP. The user can make VoIP calls remotely by dialing into the FXO line port on HT503. TO MAKE A PSTN-TO-VOIP CALL: 1. Make an incoming call to the PSTN line on FXO port. The phone will ring for 4 times by default (this setting is configurable on the configuration page). 2.
FORWARD CALLS TO PSTN Any VOIP call may be forwarded to a specified PSTN number if the call is not answered after a pre configured numbers of rings. By default “Number of Rings” parameter has value 4. For example, if the end-user has configured a cell phone number in the field “Forward to PSTN” under BASIC SETTINGS configuration page, all calls will be forwarded to the cell phone number after 4 rings. FORWARD CALLS TO VOIP By default, each incoming PSTN call is received over the FXS port.
CALL FEATURES TABLE 5: HT503 CALL FEATURE DEFINITIONS Key Call Features *30 Block Caller ID (for all subsequent calls) *31 Send Caller ID (for all subsequent calls) *47 *50 Direct IP Calling. Dial “*47” + “IP address”. No dial tone is played in the middle. Detail see Direct IP Calling section on page 12. Disable Call Waiting (for all subsequent calls) *51 Enable Call Waiting (for all subsequent calls) *67 Block Caller ID (per call). Dial “*67” + ” number ”. No dial tone is played in the middle.
CONFIGURATION GUIDE CONFIGURING HT503 THROUGH VOICE PROMPT DHCP MODE Follow Table 3 with voice menu option 01 to enable HT503 to use DHCP. STATIC IP MODE Follow Table 3 with voice menu option 01 to enable HT503 to use STATIC IP mode, then use option 02, 03, 04 to set up HT503’s IP, Subnet Mask, Gateway respectively. TFTP SERVER ADDRESS Follow Table 3 with voice menu option 06 to configure the IP address of the TFTP server.
CONFIGURING HT503 WITH WEB BROWSER HT503 ATA has an embedded Web server that will respond to HTTP GET/POST requests. It also has embedded HTML pages that allow users to configure the HT503 through a Web browser such as Microsoft’s IE, AOL’s Netscape or Mozilla Firefox installed on Windows or Unix OS. (Macintosh OS is not included). Access the Web Configuration Menu The HT503 HTML configuration page can be accessed via LAN or WAN ports. • FROM THE LAN PORT: 1. Directly connect a computer to the LAN port 2.
FIGURE 3: SCREENSHOT OF CONFIGURATION LOG-IN PAGE The password is case sensitive with maximum length of 25 characters. The factory default password for End User and administrator is “123” and “admin” respectively. Only an administrator can access the “ADVANCED SETTING” configuration page. NOTE: If you can not log into the configuration page by using the default password, please check with the VoIP service provider.
End User Password This contains the password for end user to access the Web Configuration Menu. User can put new password here. This field is case sensitive with maximum of 25 characters Web Port This is the device’s internal HTTP server port. Default is 80. Telnet Server Default is set to YES IP Address • 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.
Range: 0 – 3600, default is 300 Uplink Bandwidth 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. For example if 64 is configured, there will be at least 64kbps reserved for VoIP. The primary function of this setting is to reserve bandwidth for VoIP. Downlink Bandwidth The maximum downlink bandwidth permitted by the device. This function is disabled by default.
ADVANCED CONFIGURATION AND FXS/FXO PORTS PARAMETERS To login to the Advanced Setting and FXS port configuration pages, administrator password is required. The default administrator password is “admin”. User can change the administrator password here. The password is case sensitive and the maximum length is 25 characters.
sounds. ON is the period of ringing (“On time” in ‘ms’) while OFF is the period of silence. In order to set a continuous tone, OFF should be zero. Otherwise it will ring ON ms and a pause of OFF ms and then repeat the pattern. Lock Keypad Update Disable Voice Prompt Disable Direct IP Calling NTP server Syslog Server Syslog Level Here is an example for the configuration of the dial tone for North America: f1=350@-13,f2=440@-13,c=0/0; Syntax: f1=freq@vol, f2=freq@vol, c=on1/off1-on2/off2-on3/off3; [...
Authenticate ID Authentication Password Name Use DNS SRV: User ID is Phone Number SIP Registration Unregister on Reboot Outgoing Call w/o Registration Register Expiration Local SIP port Local RTP port Use Random Port Refer to Use Target DTMF Payload Type DTMF in Audio DTMF Via RFC2833 DTMF Via SIP INFO Send Flash Event Enable Call Features Offhook Auto-Dial Proxy-Require Use NAT IP Distinctive Ring Tone Disable Call Waiting Disable Call Waiting Tone Ring Timeout No Key Entry Timeout Early Dial Grandstr
Dial Plan Prefix Use # as Dial key Dial Plan 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. 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).
Symmetric RTP Fax Mode Fax Detection Mode Jitter Buffer Type Jitter Buffer Length SRTP Mode SLIC Setting Called ID Scheme Caller ID TX Level (dB) Polarity Reversal Loop Current Disconnect Loop Current Disconnect Duration Hook Flash Timing Gain Call Progress/ Ring Tones by preventing the transmission of “silent packets” over the network. Default is “No.
Authenticate Password SIP service subscriber’s account password. Name SIP service subscriber’s name for Caller ID display. Use DNS SRV Default is “No.” If set to “Yes” the client will use DNS SRV to look up the server. User ID is Phone Number If the HT503 has an assigned PSTN telephone number, this field should be set to “Yes”. Otherwise, set it to “No”. If “Yes” is set, a “user=phone” parameter will be attached to the “From” header in SIP request.
Dial Plan Prefix Sets the prefix added to each dialed number. Use # as Dial Key This 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”, the “#” key can be included as part of a number. Dian Plan Dial Plan Rules: 1. Accept Digits: 1,2,3,4,5,6,7,8,9,0 2. Grammar: x - any digit from 0-9; f. xx+ - at least 2 digit number; g. ^ - exclude; h.
by preventing the transmission of “silent packets” over the network. Symmetric RTP Default is “No.” When set to “Yes” the device will change the destination to send RTP packets to the source IP address and port of the inbound RTP packet last received by the device. Fax Mode T.38 (Auto Detect) FoIP by default, or fax Pass-Through (must use PCMU/PCMA) Fax Tone Detection Mode Default is Callee. This decides whether Caller or Callee sends out the re-invite for T.38 or Fax Pass-Through.
larger than the delay required to complete the PSTN caller ID delivery. DTMF Dial Pause (ms) Dial pause is the time between 2 digits for the same scenario as explained above. First Digit Timeout (sec) Used for PSTN to VoIP calls. PSTN users need to enter the FIRST digit within the first digit timeout period. Otherwise the call will be dropped. Inter Digit Timeout When dialing from the PSTN to VoIP, subsequent digits have to be input within the period of inter-digit timeout.
SAVING THE CONFIGURATION CHANGES Once a change is made, users should click on the “Update” button in the Configuration page. The HT503 will display a confirmation screen to confirm that the changes have been saved. Click ‘Reboot’ to save all changes. Please reference the GUI pages using the following link: http://www.grandstream.com/user_manuals/GUI/GUI_HT503.rar.
CONFIGURATION THROUGH A CENTRAL SERVER The Grandstream HT503 can be automatically configured from a central provisioning system. When the HT503 boots up, it will send TFTP or HTTP request to download configuration file, “cfg000b82xxxxxx”, where “000b82xxxxxx” is the LAN side MAC address of the HT503 The configuration files can be downloaded via TFTP or HTTP from the central server.
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 To upgrade via TFTP or HTTP, the “Firmware Upgrade and Provisioning upgrade via” field needs to be set to TFTP or HTTP, respectively. “Firmware Server Path” needs to be set to a valid URL of a TFTP or HTTP server, server name can be in either FQDN or IP address format. Here are examples of some valid URL. e.g.
CONFIGURATION FILE DOWNLOAD Grandstream SIP Device can be configured via Web Interface as well as via Configuration File through TFTP or HTTP. “Config Server Path” is the TFTP or HTTP server path for configuration file. It needs to be set to a valid URL, either in FQDN or IP address format. The “Config Server Path” can be same or different from the “Firmware Server Path”. A configuration parameter is associated with each particular field in the web configuration page.
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 IVR Command Reset default factory settings using the IVR Prompt (Table 5): 1. Dial “***” for voice prompt. 2.
