Grandstream Networks, Inc.
HT503 FXS/FXO Port Analog Telephone Adaptor
HT503 User Manual Firmware Version 1.0.1.63
www.grandstream.com support@grandstream.com

TABLE OF CONTENTS

HT503 USER MANUAL WELCOME..... 4
Safety Compliances.... 4 Warranty.... 4
CONNECT YOUR HT503..... 5
Equipment Packaging..... 5 Connecting the HT503.... 5

PRODUCT OVERVIEW..... 7

Software Features Overview... 7 Hardware Specification.... 8

BASIC OPERATIONS..... 9

Understanding HT503 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 Blind Transfer.... 11 Attended Transfer..... 12 3-way Conferencing.... 12 PSTN Pass Through.... 12 VoIP-to-PSTN Calls..... 13 PSTN-to-VoIP Calls..... 13 Route Calls to PSTN..... 14 Forward Calls to PSTN.... 14 Forward Calls to VoIP.... 15 One Stage Dialing.... 15 Fax Support.... 15
CALL FEATURES.... 16 CONFIGURATION GUIDE.... 17
Configuring HT503 through Voice Prompt... 17 Configuring HT503 with Web Browser... 18 Access the Web Configuration Menu... 18 Saving the Configuration Changes... 34 Rebooting from Remote... 34 Configuration through a Central Server... 34

SOFTWARE UPGRADE..... 35

Firmware Upgrade through TFTP/HTTP/HTTPS... 35 Configuration File Download.... 36 Firmware and Configuration File Prefix and Postfix... 36 Managing Firmware and Configuration File Download... 36
RESTORE FACTORY DEFAULT SETTING.... 37
HT503 User Manual Firmware 1.0.1.63
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TABLE OF FIGURES

HT503 USER MANUAL Figure 1: Connecting the HT503.... 5 Figure 2: Interconnection Diagram of the HT503... 6 Figure 3: Uplink/Downlink Bandwidth Limitation... 22

TABLE OF TABLES

HT503 USER MANUAL Table 1: Definitions of the HT503 Connectors... 6 Table 2: HT503 LED Definitions.... 6 Table 3: HT503 Technical Specifications... 7 Table 4: HT503 Hardware Specification.... 8 Table 5: HT503 IVR Menu Definitions.... 9 Table 6: HT503 Call Feature Definitions.... 16 Table 7: Status Page.... 19 Table 8: Basic Settings..... 20 Table 9: Advanced Settings.... 22 Table 10: FXS PORT Settings.... 24 Table 11: FXO PORT Settings.... 29

TABLE OF GUI INTERFACES

HT503 USER MANUAL (http://www.grandstream.com/support/ht_series/ht503/documents/ht503_gui.zip)
1. 2. 3. 4. 5. 6. 7. 8. 9.
SCREENSHOT OF CONFIGURATION LOGIN PAGE SCREENSHOT OF STATUS PAGE SCREENSHOT OF BASIC SETTINGS CONFIGURATION PAGE SCREENSHOT OF ADVANCED SETTINGS CONFIGURATION PAGE SCREENSHOT OF FXS ACCOUNT CONFIGURATION SCREENSHOT OF FXO ACCOUNT CONFIGURATION SCREENSHOT OF CALL PROGRESS TONES CONFIGURATION PAGE SCREENSHOT OF SAVED CONFIGURATION CHANGES SCREENSHOT OF REBOOT PAGE
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WELCOME

Thank you for purchasing Grandstreams 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. It enables remote call origination and termination from/to PSTN and supports the feature of hop-on/hop-off calling. This manual will help you learn how to operate and manage your HT503 Analog Telephone Adaptor/IAD and make the best use of its many upgraded features including simple and quick installation, 3-way conferencing, and remote call origination and hop-on/hop-off calling using the programmable PSTN FXO port. This HT503 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 remote user. This document is subject to changes without notice. The latest electronic version of this user manual can be downloaded from the following location: http://www.grandstream.com/support/ht_series/ht503/documents/ht503_usermanual_english.pdf

HT503 Front View HT503 Back View

Display LEDs (Green)

RJ-45 Ports 10/100 Mbps
Power Supply (12V) RJ11 RJ11 FXS Port FXO Port
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. Telephone calls can be originated from or terminated on the PSTN network remotely via the FXO port.
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TABLE 1: DEFINITIONS OF THE HT503 CONNECTORS 12VDC, 0.5A LAN Port (RJ-45) WAN Port (RJ-45) PHONE (RJ-11) LINE (RJ-11) Power adapter connection Connect the LAN port with an Ethernet cable to your PC. Connect the WAN port to the internal LAN network or router. FXS port to be connected to analog phones / fax machines. FXO port should be connected to the PSTN line
TABLE 2: HT503 LED DEFINITIONS
LEDs POWER LED WAN LED LAN LED PHONE/ LINE LED Indicates Power. Remains ON when power is connected Indicates LAN (or WAN) port activity Indicates PC (or LAN) port activity 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.
FIGURE 2: INTERCONNECTION DIAGRAM OF THE HT503
Internet ADSL/Cable Modem Ethernet Analog Phone FXS

WAN FXO

PSTN Cloud

Cordless

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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. It functions as a true FXO gateway that enables remote call origination and termination from/to PSTN and supports the feature of hop-on/hop-off using the programmable FXO port.
Software Features Overview
The HT503 features 2 SIP account profiles and supports advanced telephony features including caller ID, call waiting, call transfer, 3-way conferencing (with either IP or PSTN calls), and multi-language voice prompts. From a technical standpoint, the HT503 offers a power-outage survivable life line and internetdisconnect survivable fail-over-to-PSTN support, dual 10/100Mbps Ethernet ports with integrated highperformance NAT router, a flexible dial plan and a broad range of popular voice codecs. TABLE 3: HT503 TECHNICAL SPECIFICATIONS

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. To dial a PSTN number such as 6266667890, you may need a prefix number followed by the phone number. Please check with your VoIP service provider for this information. If your phone is assigned a PSTN-like number such as 6265556789, you will most likely follow the rule 1 + (the number) 16266667890. Press # or wait for 4 seconds.
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: Both HT503 and other VoIP Device, have public IP addresses, or Both HT503 and other VoIP Device are on the same LAN using private IP addresses, or Both HT503 and other VoIP Device can be connected through a router using public or private IP addresses (with necessary port forwarding or DMZ).
HT503 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 using format ex. 192*168*0*160 after the dial tone. Using Star Code
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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. 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 can not make direct IP calls between FXS1 to FXS2 since they are using same IP.

Call Hold

This function is applicable on the FXS port for VoIP calls only. While in conversation, pressing the flash button on the connected phone (if the phone has that button) places the remote end on hold. Pressing the flash button again releases the previously held party and the conversation can resume. If no flash button is available, then on-off hook quickly (hook flash) will do the same thing. You may lose the call if hook flash is not quick enough.

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).
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2. If no one answers the call after 4 rings (default configuration), then the caller hears either a special continuous tone (prompting a PIN number) or a dial tone. 3. Enter a valid PIN (if configured under the BASIC configuration page). The caller will hear dial tone and be bridged to VoIP. If an incorrect PIN is input, the continuous tone prompts caller to enter a valid PIN. The caller may try 3 times to enter a valid PIN, if it is invalid the HT503 will hang up. 4. The caller can dial a VoIP number followed by # (or wait for 4 seconds); the VoIP call will be initiated from the SIP account configured on the FXO port. NOTE: Users can choose whether or not to apply password protection for VoIP-to-PSTN calls. A PIN (Pin for PSTN calls) consists of up to 8 numeric digits and can be configured using the BASIC SETTINGS of the web configuration page. By default, there is no password protection. (I.e. there is no authentication required for callers on the use of PSTN line through HT503). When a PIN is configured for VOIP-to-PSTN call flow, the VoIP device that calls into the HT503 FXO account needs to configure RFC2833 or SIP Info for DTMF digit transmission. The special continuous tone is the prompt to enter a valid PIN code. If a caller doesnt enter a valid PIN, the HT503 times out after 10 seconds. Users may press the # key to indicate the end of an input or wait 4 seconds. On the web configuration page, if the Forward to VoIP is configured, the second stage dialing format is eliminated, so after dialing into the FXO SIP account number, the PSTN number will be called automatically

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. If the dialed digits match the specified prefix, outbound calls will be initiated from the PSTN line. Note: The route to PSTN feature is only applicable to a phone connected to the FXS Port. The configuration is done using the dial plan feature under the FXS tab. An example of the configuration is {L: 911x+}. This shows that only calls that start with 911 are immediately forwarded to the PSTN line. All other numbers will not be routed to the PSTN. An normal # would be: {L: 617x+|x+} or {x+| L: 617x+} For example, if Route Call to PSTN is configured as {L: 626x+}, all outgoing calls starting with 626 will be initiated from the PSTN line.

*03 *30 *31 *47 *50 *51 *67 *82 *69 *70 *71 *72 *73 *78 *79 *87 *90 *91 *92 *93 Flash/Hook #
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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 HT503s IP, Subnet Mask, Gateway respectively. FIRMWARE SERVER IP ADDRESS Select voice menu option 13 to configure the IP address of the firmware server. CONFIGURATION SERVER IP ADDRESS Select voice menu option 14 to configure the IP address of the configuration server. UPGRADE PROTOCOL Select voice menu option 15 to choose firmware and configuration upgrade protocol. User can choose between TFTP, HTTP and HTTPS. FIRMWARE UPGRADE MODE Select voice menu option 17 to choose firmware upgrade mode. There are three options: 1) always check, 2) check only when pre/suffix changes, and 3) never upgrade WAN PORT WEB ACCESS Select voice menu option 12 to enable WAN Port Wed Access of the device configuration pages.
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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 Microsofts IE, AOLs 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. 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. FROM THE WAN PORT: 1. Follow table 4 to find the WAN side IP address. 2. Open a web browser, type in the WAN side IP address for example: http://HT503-WAN-IP-Address
Note: WAN side HTTP access is disabled by default for security reason. You can enable HTTP access on the configuration page by setting WAN side HTTP access to be YES. Initial access to the configuration pages is always from the LAN port. The instructions are listed above. The IVR announces 12 digits IP address, you need to strip out the leading 0 in the IP address. For ex. IP address: 192.168.001.014, you need to type in http://192.168.1.14 in the web browser.
Once the HTTP request is entered and sent from a web browser, the user will see a log-in screen. There are two default passwords for the login page: User Level: End User Level Administrator Level Password: 123 admin Web pages allowed: Only Status and Basic Settings Browse all pages

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, FXS PORT and FXO PORT configuration pages.
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.
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Only an administrator can access the ADVANCED SETTING, FXS PORT and FXO PORT configuration pages. Please reference the GUI pages using the following link: http://www.grandstream.com/support/ht_series/ht503/documents/ht503_gui.zip.

DEFINITIONS

This section will describe the options in the Web configuration user interface. As mentioned, a user can log in as an administrator or end-user. Functions available for the end-user are: STATUS: Displays the network status, account status, software version and MAC-address of the phone BASIC SETTINGS: Basic preferences such as date and time settings, multi-purpose keys and LCD settings can be set here. Additional functions available to administrators are: ADVANCED SETTINGS: To set advanced network settings, codec settings and XML configuration settings. FXS PORT: To configure the FXS port. FXO PORT: To configure the FXO port.

TABLE 7: STATUS PAGE

MAC Address The device ID, in HEX format. This is very important ID for ISP troubleshooting. Both LAN and WAN MAC addresses are located here. The LAN MAC address is used for provisioning and is written on the label in the original box as well as on the label located on the back panel of the device. This field shows IP address of the HT503. This field contains the product model info, such as HT503. Program: This is the main software release. This number is always used for firmware upgrade. Current release is 1.0.0.15 Bootloader: current version is 1.0.0.7 Core: current version 1.0.0.23 Base: current version is 1.0.0.66 This shows system up time since last reboot. This shows whether the PPPoE is up if connected to DSL modem This shows what kind of NAT the HT503 is connected to. It is based on STUN protocol. If the detected NAT is symmetric NAT, STUN will not work and Outbound Proxy needed to make HT503 functioning correctly. Displays information regarding the individual FXS ports. Port Hook Registration DND Forward FXS FXO On Hook Idle Registered Registered Yes No Busy Forward Delayed Forward

Uplink Bandwidth

Downlink Bandwidth
Enable UPnP Reply to ICMP on WAN Port WAN Side HTTP/Telnet Access Cloned WAN MAC Address LAN DHCP Base IP LAN DHCP Start IP LAN DHCP End IP LAN Subnet Mask DHCP IP Lease Time DMZ IP: Port Forwarding: PSTN access code
PIN for PSTN calls PIN for VoIP calls Unconditional Call Forward to PSTN Unconditional Call Forward to VoIP
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FIGURE 3: UPLINK/DOWNLINK BANDWIDTH LIMITATION
Advanced User configuration includes not only the end user configuration, but also advanced configurations such as: SIP configuration, Codec selection, NAT Traversal Setting and other miscellaneous configuration. TABLE 9: ADVANCED SETTINGS
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. This field defines the layer 3 QoS parameter which can be the value used for IP Precedence or Diff-Serv or MPLS. Default value is 48. Layer 2 QoS settings. Default setting is blank. VLAN supported equipment is required when configuring these settings. IP address or Domain name of the STUN server. This parameter specifies how often the HT503 sends a blank UDP packet to the SIP server in order to keep the NAT pin hole open. Default is 20 seconds. Use STUN keep-alive to detect WAN side network problems. If keep-alive request does not yield any response for configured number of times, the device will restart the TCP/IP stack. If the STUN server does not respond when the device boots up, the feature is disabled. Enables the HT503 to download firmware or configuration files through either TFTP or HTTP servers. The default method is HTTP. This is the IP address of the configured TFTP server. If this is configured, the HT503 retrieves the new configuration file or new code image from the specified TFTP server at boot time. After 5 attempts, the system will timeout and will start the boot process using the existing code image in the Flash memory. If a TFTP server is configured and a new code image is retrieved, the new downloaded image is saved into the Flash memory. Note: Firmware upgrades may take up to 10 minutes depending on your network environment. On a LAN it usually takes about 2 minutes. Please do NOT interrupt the TFTP upgrade process (especially the power supply) as this will damage the device. Depending on the network environment this process can take up to 15 or 20 minutes. Via HTTP The URL for the HTTP server used for firmware upgrade and configuration via HTTP. For example, http://provisioning.mycompany.com:6688/Grandstream/1.0.0.6 :6688 is the specific TCP port where the HTTP server is listening; Omit if using default port 80. Note: If Auto Upgrade is set to No, F/W will download at boot time. The URL of the HTTP server used for firmware upgrade and configuration via a secure HTTP connection. HT503 User Manual Firmware 1.0.1.63 Page 22 of 37 Last Updated: 11/2010

Layer 3 QoS Layer 2 QoS STUN Server Keep-alive interval Use STUN to detect network activity
Firmware Upgrade and Provisioning Via TFTP

Via HTTPS

Firmware Server Path Config Server Path Firmware File Prefix Firmware File Postfix Config File Prefix Config File Postfix Automatic Upgrade
For example, https://provisioning.mycompany.com Note: the HTTPS default port is 443. IP address or domain name of firmware server. IP address or domain name of configuration server. 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. Default is blank. End users should keep it blank. Default is blank. End users should keep it blank. Default is blank. End users should keep it blank. Choose Yes to enable automatic upgrade and provisioning. When set to No, HT503 will only do upgrade once at boot up. When Check every day or Check every week is checked, user can specify Hour of the day(0-23) or Day of the week(0-6). Default time is Monday 1AM. There are three options to choose from: Always check for New Firmware at Boot up, Check New Firmware only when F/W pre/suffix changes, and Always Skip the Firmware Check. This protects the configuration from an unauthorized change. If set to Yes, the configuration file is authenticated before acceptance. Key for firmware encryption. (32 digits in hexadecimal format. End users should keep it blank. The user specified SSL certificate used for SIP over TLS in X.509 format. The user specified SSL private key used for SIP over TLS in X.509 format. User specified password to protect the private key above. Configuration option for FXS port ring cadence for all incoming calls. (Syntax: c=on1/off1on2/off2-on3/off3; [.]) Using these settings, users can configure tone frequencies according to their preference. By default they are set to North American frequencies. These tones should be configured with known values to avoid uncomfortable high pitch sounds. ON is the period of ringing (On time in ms) while OFF is the period of silence. In order to set a continuous tone, OFF should be zero. Otherwise it will ring ON ms and a pause of OFF ms and then repeat the pattern. Example for North America Dial Plan: f1=350@-13,f2=440@-13,c=0/0; Syntax: f1=freq@vol, f2=freq@vol, c=on1/off1-on2/off2-on3/off3; [.] (Note: freq: 0 - 4000Hz; vol: -30 - 0dBm) If set to Yes, the configuration update via keypad is disabled. Note: some informative options still will be available for users after configuring to Yes. Changing existing configuration will be impossible. Disables the voice prompt configuration. Default is No. If set to Yes accessing integrated voice menu will be impossible. Disables the Direct IP Call function. Default is No. If set to Yes to make direct IP call will be impossible. Life line feature ensures user can place/receive a PSTN call in an emergency situation. 1. If set to Auto, in case of power loss or loss of SIP registration, the PSTN line will be seamlessly connected to analog phone connected to FXS port. 2. If set to Always Connected the PSTN line will be always connected to the phone connected to FXS port. VoIP calls will not be allowed in this configuration. 3. If set to Always Disconnected, user can only place VoIP calls, regardless of any power loss and/or SIP registration problems. User will be unable to place/receive any PSTN calls. URL or IP address of the NTP server, Used to synchronize the date/time. The IP address or URL of syslog server, especially useful for ITSP HT503 User Manual Firmware 1.0.1.63 Page 23 of 37 Last Updated: 11/2010

Proxy-Require Use NAT IP Distinctive Ring Tone
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 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 Massachusetts. Any other incoming call will ring using cadence defined in parameter System Ring Cadence located under Advanced Settings Configuration page. Default is No. Default is No. This is to disable the caller ID when a call waiting information arrives. 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. Sets the time in which an incoming call will stop ringing when not picked up. 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. 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. 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). Note: This feature is NOT designed to work with and should NOT be enabled for direct IP-to-IP calling. Sets the prefix added to each dialed number. 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. Dial Plan Rules: 1. 2. Accept Digits: 1,2,3,4,5,6,7,8,9,0 , *, #, A,a,B,b,C,c,D,d Grammar: x - any digit from 0-9; a. xx+ - at least 2 digits number; b. xx. at least 2 digits number; c. ^ - exclude; d. [3-5] - any digit of 3, 4, or 5; e. [147] - any digit 1, 4, or 7; f. <2=011> - replace digit 2 with 011 when dialing Example 1: {[369]11 | 1617xxxxxxx} Allow 311, 611, 911, and any 10 digit numbers of 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 length of number with leading digit 2 and 10 digit-numbers of leading digit 1 and leading exchange number between 2 and 9; If leading digit is 2, HT503 User Manual Firmware 1.0.1.63 Page 26 of 37 Last Updated: 11/2010

User ID is Phone Number

SIP Registration Unregister on Reboot Outgoing Call Without Registration Register Expiration
SIP registration failure retry wait time Local SIP Port Local RTP Port
When configured, the FXO port will use this port _value for RTP and the port_value+1 for its RTCP. 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. 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. If set to Yes, the Outbound Proxy will be removed from the 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. If the network latency is high, select larger value for reliable usage. Maximum retransmission interval for non-INVITE requests and INVITE responses. Sends DTMF using RFC2833 The HT503 supports up to 3 different DTMF methods including in-audio, via RTP (RFC2833) and via Sip Info. User can configure DTMF method in a priority list. SIP Extension to notify SIP server that the unit is behind a NAT/Firewall. NAT IP address used in SIP/SDP message. Default is blank. Sets the time in which an incoming from PSTN call will stop ringing when not picked up. 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. 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). Note: This feature is NOT designed to work with and should NOT be enabled for direct IP-to-IP calling. Dial Plan Prefix Use # as Dial Key Sets the prefix added to each dialed number. 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. 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. In case of normal dialing to VoIP, after dialing PSTN number, If using the hop on/hop off feature, the dial plan rules affect only the last called number (i.e. the number called after receiving dial tone from the ATA). Dial Plan Rules: 4. 5. Accept Digits: 1,2,3,4,5,6,7,8,9,0 , *, #, A,a,B,b,C,c,D,d Grammar: x - any digit from 0-9; a. b. c. d. Grandstream Networks, Inc. xx+ - at least 2 digits number; xx. at least 2 digits number; ^ - exclude; [3-5] - any digit of 3, 4, or 5; HT503 User Manual Firmware 1.0.1.63 Page 30 of 37 Last Updated: 11/2010

Refer to Use Target Contact Remove OBP from Route Header Support SIP instance ID Validate incoming message Check SIP User ID for incoming INVITE SIP T1 Timeout SIP T2 Interval DTMF Payload Type Preferred DTMF method (in listed order) Proxy Require Use NAT IP Ring Timeout Early Dial

Dian Plan

e. [147] - any digit 1, 4, or 7; f. <2=011> - replace digit 2 with 011 when dialing Example 1: {[369]11 | 1617xxxxxxx} Allow 311, 611, 911, and any 10 digit numbers of 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 length of number with leading digit 2 and 10 digit-numbers of leading digit 1 and leading exchange number between 2 and 9; If leading digit is 2, replace leading digit 2 with 011 before dialing. 6. 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 will be: { [x*]+ }. Subscribe for MWI Anonymous Call Rejection Special Feature Session Expiration Default is No. When set to Yes a SUBSCRIBE for Message Waiting Indication will be sent periodically. Default is No. If set to Yes, incoming calls with anonymous Caller ID will be rejected with a 486 busy message. Default is Standard. Choose the selection to meet some special requirements from Softswitch vendors. Grandstream implemented SIP Session Timer. The session timer extension enables SIP sessions to be periodically refreshed via a SIP request (UPDATE, or re-INVITE. Once the session interval expires, if there is no refresh via a UPDATE or re-INVITE message, the session will be terminated. Session Expiration is the time (in seconds) at which the session is considered timed out, if no successful session refresh transaction occurs beforehand. The default value is 180 seconds. The minimum session expiration (in seconds). The default value is 90 seconds. If selecting Yes the phone will use session timer when it makes outbound calls if remote party supports session timer. If selecting Yes the phone will use session timer when it receives inbound calls with session timer request. If selecting Yes the phone will use session timer even if the remote party does not support this feature. Selecting No will allow the phone to enable session timer only when the remote party support this feature. To turn off Session Timer, select No for Caller Request Timer, Callee Request Timer, and Force Timer. As a Caller, select UAC to use the phone as the refresher, or UAS to use the Callee or proxy server as the refresher. HT503 User Manual Firmware 1.0.1.63 Page 31 of 37 Last Updated: 11/2010

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SOFTWARE UPGRADE

Software upgrade can be done via either 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. Here are examples of some valid URL. e.g. firmware.mycompany.com:6688/Grandstream/1.0.1.63 e.g. 72.172.83.110 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 HTTP server. Its address can be found at http://www.grandstream.com/firmware.html. Currently the HTTP firmware server IP address is 72.172.83.110. 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 HT503 will attempt to retrieve the new image files by downloading them into the HT503 s SRAM. During this stage, the HT503s 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 HT503 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 HTTP 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. 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 HT503 device in the same LAN segment.

Grandstream Networks, Inc. HT503 User Manual Firmware 1.0.1.63 Page 35 of 37 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 side 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 possible to store ALL of the firmwares 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 selected, 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 to Yes, Service Provider can use P193 to have the devices periodically check with either Firmware Server or Config Server, whenever they are defined. This allows the device periodically check whether there is any new changes need to be taken, similar to the AntiVirus Software to upgrade the Virus Definition files. Screenshot is below: