Grandstream Networks, Inc.
HT503 USER MANUAL INDEX GNU GPL INFORMATION .......................................................................... 5 CHANGE LOG ........................................................................................... 6 CHANGES FROM 1.0.7.6 USER MANUAL .......................................................................................... 6 CHANGES FROM 1.0.6.8 USER MANUAL .......................................................................................... 6 WELCOME .........................
CALL FEATURES .................................................................................... 25 CONFIGURATION GUIDE........................................................................ 27 CONFIGURING HT503 THROUGH VOICE PROMPT ....................................................................... 27 CONFIGURING HT503 WITH WEB BROWSER ................................................................................ 27 ACCESS THE WEB CONFIGURATION MENU ................................................
TABLE OF FIGURES HT503 User Manual Figure 1: CONNECTING THE HT503 ............................................................................................ 10 Figure 2: INTERCONNECTION DIAGRAM OF THE HT503 ......................................................... 11 Figure 3: UPLINK/DOWNLINK BANDWIDTH LIMITATION ........................................................... 33 TABLE OF TABLES HT503 User Manual Table 1: DEFINITIONS OF THE HT503 CONNECTORS ....................................................
GNU GPL INFORMATION HT503 firmware contains third-party software licensed under the GNU General Public License (GPL). Grandstream uses software under the specific terms of the GPL. Please see the GNU General Public License (GPL) for the exact terms and conditions of the license. Grandstream GNU GPL related source code can be downloaded from Grandstream web site from: http://www.grandstream.com/support/faq/gnu_gpl . FIRMWARE VERSION 1.0.9.
CHANGE LOG This section documents significant changes from previous versions of HT503 user manuals. Only major new features or major document updates are listed here. Minor updates for corrections or editing are not documented here. CHANGES FROM 1.0.7.6 USER MANUAL Added option to enable/disable SIP NOTIFY Authentication. [Error! Reference source not found.] Added option [Use Configured IP ] in DNS mode.
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.
your reference. This document is subject to change without notice. The latest electronic version of this user manual is available for download from the following location: http://www.grandstream.com/products/ht_series/ht503/documents/ht503_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 without the express written permission of Grandstream Networks, Inc. is not permitted. FIRMWARE VERSION 1.0.9.
CONNECT YOUR HT503 EQUIPMENT PACKAGING The HT503 ATA package contains: One HT503 Main Case One Universal Power Adaptor One Ethernet Cable One HT503 Vertical Stand CONNECTING 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.
HT503 Back View HT503 Front View Display LEDs (Green) RJ-45 Ports 10/100 Mbps Reset Power Supply (12V) RJ11 RJ11 FXS Port FXO Port FIGURE 1: CONNECTING THE HT503 The HT503 has one FXS port and one FXO port. The PHONE port next to the power supply is an FXS port. The LINE port on the back right of the HT503 is an FXO port. Both the FXS port and the FXO port can have a separate SIP account. This is a key feature of HT503 as it supports simultaneous calls on both the FXS port and FXO port.
TABLE 2: HT503 LED DEFINITIONS POWER LED Indicates Power. Remains ON when power is connected WAN LED Indicates LAN (or WAN) port activity LAN LED Indicates PC (or LAN) port activity PHONE/ LINE LED Indicates the status of the FXS and FXO ports on the back panel. Busy – ON (Solid Green) Available – OFF Slow blinking FXS LEDs indicates voicemail for that port. Note: Slow blinking of POWER, WAN, and LAN LEDs together indicate firmware upgrade/provisioning state.
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.
DHCP Server/Client Yes Audio Features Advanced Digital Signal Processing (DSP) Dynamic negotiation of codec and voice payload length Support for G.723, G.729/E, G.711, G.726-40/32/24/16, iLBC, T.38 codecs In-band and out-of-band DTMF (in audio, RFC2833, SIP INFO) Silence Suppression, VAD (voice activity detection), CNG (comfort noise generation), ANG (automatic gain control) Adaptive jitter buffer control Packet delay & loss concealment (PLC) & G.
HARDWARE SPECIFICATION The table below lists the hardware specification of HT503. TABLE 4: 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 1x PSTN pass-through and life line port telephone port (PSTN Port) LED Power, WAN, LAN, PHONE, and LINE (Green) Universal Switching Input: 100–240 VAC, 50-60 Hz Power Adaptor Output: 12VDC, 0.
BASIC OPERATIONS UNDERSTANDING HT503 VOICE PROMPT HT503 has a built-in voice prompt menu for simple device configuration. The voice prompt menu is designed for the FXS port only. To enter the voice prompt menu, press *** from the analog phone connected to the FXS port.
13 Firmware Server IP Address Announces current Firmware Server IP address. Enter 12 digit new IP address. 14 15 Configuration Server IP Announces current Config Server Path IP address. Enter 12 digit Address new IP address. Upgrade Protocol Upgrade protocol for firmware and configuration update. Press “9” to toggle between TFTP / HTTP / HTTPS 16 Firmware Version Firmware version information. 17 Firmware Upgrade Firmware upgrade mode.
PLACING A PHONE CALL PHONE OR EXTENSION NUMBERS There are currently two methods to make an extension number call: a) Dial the numbers directly and wait for 4 (default) seconds. b) Dial the numbers directly, and press # (assuming that “use # as dial key” is selected in the web configuration). Examples: To dial another extension on the same proxy, such as 1008, simply pick up the attached phone, dial 1008 and then press the # or wait for 4 seconds.
Using Star Code 1. Pick up the analog phone then dial “*47” 2. Enter the target IP address using same format as above. Note: NO dial tone will be played between step 1 and 2. Destination ports can be specified by using “*” (encoding for “:”) followed by the port number. Examples: 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.
CALL TRANSFER The HT503 supports both blind transfer and attended transfer. BLIND TRANSFER This function is applicable using the FXS port for VoIP calls only. Assume that parties A and B are in conversation. Party A wants to Blind Transfer Party B to C: 3. A presses FLASH on the analog phone to hear the dial tone. 4. Then A dials *87, then dials C’s number, and then presses # 5. A can hang up. NOTE: “Enable Call Feature” has to be set to “Yes” in web configuration page.
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. 6. If A hangs up, the conference will be terminated for all three parties when configuration “Transfer on Conference Hangup” is set to “No”.
VOIP-TO-PSTN CALLS This function is available using the FXO port. The FXO port functions as a bridge between the Internet and PSTN. The user can remotely use a PSTN line to initiate a call. TO MAKE A VOIP-TO-PSTN CALL: 1. Dial the FXO SIP account phone number to establish the VoIP session. The caller will hear the ring back tone once. Then the caller hears either a special continuous tone or a dial tone.
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 FXO port configuration page). 2.
ROUTE CALLS TO PSTN The FXO port enables access to the PSTN network. By default, the HT503 is in VoIP mode at off-hook. If “Route Call to PSTN” is configured, certain calls will be initiated from the FXO PSTN line port. This call feature is especially useful for emergency calls or local telephone calls. To use this feature, users need to specify a special rule using the dial plan parameter located under FXS Port configuration page.
can also be found under BASIC SETTINGS configuration page. ONE STAGE DIALING This feature is applicable for VoIP to PSTN calls. Any VoIP extension may dial directly to a local PSTN number if the one-stage dialing feature is activated. This feature is configured under the FXO Configuration page and requires SIP Server configuration and support. The special dial plan feature must be activated in the SIP Server.
CALL FEATURES TABLE 6: HT503 CALL FEATURE DEFINITIONS Key Call Features *02 Forcing a Codec (per call) *027110 (PCMU), *027111 (PCMA), *02723 (G723), *02729 (G729), *0272616 (G726-r16), *0272624 (G724-r24), *0272632 (G726-r32), *0272640 (G726-r40), *027201 (iLBC) *03 Disable LEC (pe call) Dial “*03” + ” number ”. No dial tone is played in the middle.
Flash/Hook Toggles between active call and incoming call (call waiting tone). If not in conversation, flash/hook will switch to a new channel for a new call. # Pressing pound sign will server as Re-Dial key. FIRMWARE VERSION 1.0.9.
CONFIGURATION GUIDE CONFIGURING HT503 THROUGH VOICE PROMPT DHCP MODE Follow Table 4 with voice menu option 01 to enable HT503 to use DHCP. STATIC IP MODE Follow Table 4 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. FIRMWARE SERVER IP ADDRESS Select voice menu option 13 to configure the IP address of the firmware server.
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. Open a command window on the computer 3. Type in “ipconfig /release”, the IP address etc becomes 0 4. Type in “ipconfig /renew”, the computer gets an IP address in 192.168.2.x segment by default 5. Open a web browser, type in the default IP address of the LAN port. http://192.168.2.1. You will see the log in page of the device.
End User and administrator is “123” and “admin” respectively. Only an administrator can access the “ADVANCED SETTING”, “FXS PORT” and “FXO PORT” configuration pages. NOTE: If you cannot 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.
Product Model This field contains the product model info, such as HT503. Software Version Program: This is the main software release. This number is always used for firmware upgrade. Current release is 1.0.7.6 Boot and Loader are seldom changed. Bootloader: current version is 1.0.0.9. Core: current version 1.0.7.1 Base: current version is 1.0.7.6 CPE: current version is 1.0.1.19 System Uptime This shows system up time since last reboot.
• If Static IP mode is selected, the IP address, Subnet Mask, Default Router IP address, DNS Server 1 (mandatory), DNS Server 2 (optional) fields need to be configured. DHCP hostname This option specifies the name of the client. This field is optional but may be required by some Internet Service Providers. Default is blank. DHCP vendor class ID This option is used by clients and servers to exchange vendor-specific information. Default is blank. PPPoE account ID PPPoE username.
Range: 0 - 3600 NAT UDP Timeout NAT TCP idle timeout in seconds. Connection will be closed after preconfigured, timeout if not refreshed. 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, 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.
Port Forwarding: Allows users to forward a matching (TCP/UDP) port to a specific LAN IP address with a specific (TCP/UDP) port. The code to access the PSTN line (Maximum 5 digits). Default is “*00”. Any time user PSTN access code can make PSTN calls from the analog phone connected to FXS port. By default, user may pick up the phone, dial *00, and after obtaining PSTN line ( user will hear regular dial tone) normal PSTN dialing is allowed.
Admin Password Administrator password. Only the administrator can configure the “Advanced Settings” page. Password field is purposely blanked for security reason after clicking update and saved. The maximum password length is 25 characters. Layer 3 QoS 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 Layer 2 QoS settings. Default setting is blank.
XML Config File The password used for encrypting the XML configuration file using OpenSSL. Password This is required for the phone to decrypt the encrypted XML configuration file. HTTP/HTTPS User Name The user name for the HTTP/HTTPS server. HTTP/HTTPS Password The password for the HTTP/HTTPS server. Firmware File Prefix Default is blank. If configured, HT503 will request the firmware file with the prefix. This setting is useful for ITSPs. End user should keep it blank.
CPE SSL Private Key Configure the SSL Private Key of Customer-premises equipment System Ring Cadence Configuration option for FXS port ring cadence for all incoming calls. (Syntax: c=on1/off1-on2/off2-on3/off3; ) Note : Maximum supported cadences is 3 Call Progress Tones Using these settings, users can configure tone frequencies according to their preference. By default they are set to North American frequencies.
Syslog Level Select the ATA to report the log level. Default is NONE. The level is either one of DEBUG, INFO, WARNING or ERROR.
provide these settings. If this field is set to “Yes”, then the device will periodically send a dummy UDP packet to the SIP server to pinhole the NAT. SIP User ID User account information, provided by VoIP service provider (ITSP), usually has the form of digit similar to phone number or actually a phone number. This field contains the user part of the SIP address for this phone. e.g., if the SIP address is sip:my_user_id@my_provider.com, then the SIP User ID is: my_user_id.
Reregister before This parameter allows the user to specify the reregisteration time before expiration. Expiration Local SIP port This parameter defines the local SIP port the HT503 will listen and transmit. The default value for FXS port is 5060. Local RTP port This parameter defines the local RTP port pair used by the HandyTone ATA. The default value for FXS port is 5004. Use Random Port Default is No. If set to Yes, the device will pick randomly-generated SIP and RTP ports.
Note: User will need this IP address when accessing the IVR via the web configuration page. Offhook Auto-Dial Delay Configure the delay time for offhook auto-dial function. Range is 0-60 seconds, default is 0. Proxy-Require SIP Extension to notify SIP server that the unit is behind the NAT/Firewall. Use NAT IP NAT IP address used in SIP/SDP message.
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).
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.
UAC Specify Refresher As a Caller, select UAC to use the phone as the refresher, or UAS to use the Callee or proxy server as the refresher. UAS Specify Refresher As a Callee, select UAC to use caller or proxy server as the refresher, or UAS to use the phone as the refresher. Send Re-INVITE After Default is No, If set to “Yes”, device will send an INVITE with audio vocoders upon Fax completition of Fax to continue session in audio only.
G723 Rate This defines the encoding rate for G723 vocoder. Default setting is 6.3kbps. iLBC Frame Size This sets the iLBC size in 20ms or 30ms iLBC Payload Type This defines payload type for iLBC. Default value is 97. The valid range is between 96 and 127. AAL2-G726-16 Payload Defines payload type for AAL2-G726-16. Default value is 100. Range is from 96 to Type 127. AAL2-G726-24 Payload Defines payload type for AAL2-G726-24. Default value is 99. Range is from 96 to 127.
Duration indicate to the local party that the call is disconnected from the remote side. (100-10000 ms. Default 200 ms) Enable Hook Flash Default is Yes. If set to “No”, FLASH button could only be used for terminating calls. Hook Flash Timing The time period when the cradle is pressed (Hook Flash) to simulate a FLASH. Adjust this time value to prevent unwanted activation of the Flash/Hold and automatic phone ring-back.
NAT Traversal (STUN) This parameter defines whether or not the HT503 NAT traversal mechanism is activated. If set to “Yes” with a STUN server also specified, the HT503 will perform according to the STUN client specification. Using this mode, the embedded STUN client will detect if and what type of firewall/NAT is being used. If the detected NAT is a Full Cone, Restricted Cone, or a Port-Restricted Cone, the HT503 will use its mapped public IP address and port in all of its SIP and SDP messages.
Local RTP Port This parameter defines the local RTP port pair used by the HandyTone ATA. The default value for FXO port is 5012. Use Random Port This parameter forces the random generation of both the local SIP and RTP ports when set to Yes. This is usually necessary when multiple HT503 units are behind the same NAT. Refer to Use Target Default is No. If set to YES, then for Attended Transfer, the “Refer-To” header uses the Contact transferred target’s contact header information.
IP-to-IP calling. 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 plans work only for incoming calls from PSTN network. In case unconditional call forward to VoIP is configured, dial plan feature will not work.
• <=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.
the phone as the refresher. Force INVITE Session Timer can be refreshed using INVITE method or UPDATE method. Select “Yes” to use INVITE method to refresh the session timer. Invite Ring-No-Answer Default is 40 seconds, the range is between 5 and 300 seconds. Timeout Enable 100rel The use of the PRACK (Provisional Acknowledgement) method enables reliability to be offered to SIP provisional responses (1xx series). This is very important if PSTN internetworking is to be supported.
AAL2-G726-32 Payload Defines payload type for AAL2-G726-24. Default value is 104. Range is from 96 to Type 127. AAL2-G726-40 Payload Defines payload type for AAL2-G726-40. Default value is 103. Range is from 96 to Type 127. VAD Default is No. VAD allows detecting the absence of audio and conserves bandwidth by preventing the transmission of “silent packets” over the network. Symmetric RTP Default is No.
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 FXO Port Configuration page. These parameters affects call volume ONLY for calls placed to/from PSTN and VoIP networks. If call volume is too low when using VoIP extension, adjust volume using the Rx Gain Level parameter under the FXO Port Configuration page.
PSTN Ring Thru Delay If the PSTN Ring Thru Delay is set to Yes, all incoming PSTN calls through FXO will (sec) ring the phone connected to the FXS port, after this delay or after caller id is detected (whichever comes first). DTMF Digit Length (ms) Digit length and Dial Pause are port digit dialing configurations; FXO needs to dial out digits for VOIP to PSTN 1 stage calls, and unconditional call forward to PSTN, and route to PSTN. Digit Length is the play time for each digit.
CONFIGURATION THROUGH A CENTRAL SERVER Grandstream HT503 can be automatically configured from a central provisioning system. When HT503 boots up, it will send TFTP or HTTP/HTTPS requests to download configuration files, “cfg000b82xxxxxx” and “cfg00082xxxxxx.xml”, where “000b82xxxxxx” is the LAN MAC address of the HT503. If the download of “cfgxxxxxxxxxxxx.xml” is not successful, the provision program will issue request a generic configuration file “cfg.xml”.
SOFTWARE UPGRADE Software upgrade can be done via TFTP, HTTP or HTTPS. The corresponding configuration settings are in the ADVANCED SETTINGS configuration page. FIRMWARE UPGRADE THROUGH TFTP/HTTP/HTTPS To upgrade via TFTP, HTTP or 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.
environment if possible. For users who do not have a local firmware upgrade server, Grandstream provides a NAT-friendly HTTP server on the public Internet for firmware upgrade. Grandstream’s latest firmware is available http://www.grandstream.com/support/firmware. 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.
When a Grandstream device boots up or reboots, it will issue a request for a configuration file “cfgxxxxxxxxxxxx”, where “xxxxxxxxxxxx” is the MAC address of the device, i.e., “cfg000b820102ab”. In addition, device will also requests a XML configuration file “cfgxxxxxxxxxxxx.xml”. If the download of “cfgxxxxxxxxxxxx.xml” is not successful, the provision program will issue a request for a generic configuration file “cfg.xml”. Configuration file name should be in lower case letters.
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.
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.
