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1. 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 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 2 digits number A exclude 3 5 any digit of 3 4 or 5 147 any digit 1 4 or 7 lt 2 011 gt replace digit 2 with 011 when dialing e Example 1 369 11 1617xxxxxxx Allow 311 611 911 and any 10 digit numbers of leading digits 1617 e Example 2 1900x lt 1617 gt xxxxxxx Block any number of leading digits 1900 and add prefix 1617 for any dialed 7 digit numbers SOAP O Grandstream Networks Inc HT503 User Manual Firmw
2. AR Innovative IP Voice amp Video Default is Callee This decides whether Caller or Callee sends out the re invite for T 38 or Fax Pass Through Select either Fixed or Adaptive based on network conditions Select Low Medium or High based on network conditions Secure RTP protocol used for media transmission over VoIP Disabled by default Other modes are enabled but not forced amp enabled and forced Bellcore Telcordia ETSI FSK ETSI DTMF SIN 227 BT amp NTT Japan An adjustable value for the Caller ID signal to help this device to recognize Caller ID from different networks 50 OdB Default 30dB According to customer s choice CID information will be transferred from PSTN network to VoIP network using following rules 1 via SIP from PSTN CID is in the SIP From field 2 via P Asserted Identity SIP From field uses the pre configured account user Id PSTN CID is in the P Asserted Identity field 3 Send anonymous SIP From field uses anonymous PSTN CID is put in the P Asserted ldentity field 4 Disable PSTN CID will not be sent SIP From field uses the pre configured account user ID 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 Voice path volume adjustment e RX is again level for signals transmitted by FXO FXO To VoIP volume e TX is again level for signals received by FXO FXO T
3. If the dialed digits match one of the specified prefix here outbound calls will be initiated from PSTN line This field is especially useful for emergency calls Calls are unconditionally forwarded to the specified PSTN phone number for all incoming VoIP calls on FXO port Calls are unconditionally forwarded to the specified VoIP phone number for all incoming PSTN calls Each incoming call from the PSTN will first ring the analog phone connected to FXS port This call from the PSTN network will be forwarded to the preconfigured VoIP extension if it is not answered User can configure the number of rings before forwarding calls to the VoIP extension Configure number of rings using the number of rings parameter located in the FXO Port Configuration page Grandstream Networks Inc HT503 User Manual Firmware 1 0 0 9 Page 21 of 37 Last Updated 9 2007 Dag Innovative IP Voice amp Video 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 TABLE 9 HT503 ADVANCED SETTINGS PAGE DEFINITIONS Admin Password Layer 3 QoS Layer 2 QoS STUN Server Keep alive interval Firmware Upgrade and Provisioning Via TFTP Server Via HTTP Server Firmware Serv
4. Route Call to PSTN is configured as IL 626x all outgoing calls starting with 626 will be initiated from the PSTN line Grandstream Networks Inc HT503 User Manual Page 14 of 37 Firmware 1 0 0 9 Last Updated 9 2007 ndstream Innovative IP Voice amp Video FORWARD CALLS TO PSTN Any VOIP call may be forwarded to a specified PSTN number FXO port should be registered with some preconfigured number for example 1111 Any VoIP extension can dial this FXO account number and will be automatically forwarded to preconfigured PSTN extension 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 The end user may forward such a call to any preconfigured VoIP extension in case the call is not answered in a certain number of rings The Default value of the parameter Number of Rings is 4 This parameter located under FXO Port configuration page If during 4 rings the incoming from the PSTN call is not answered the call will be forwarded to another VoIP number previously configured in the field Forward to VolP This parameter 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 dia
5. 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 LAN Subnet Mask LAN DHCP Base IP DHCP IP Lease Time DMZ IP Port Forwarding PSTN access code PIN for PSTN calls PIN for VoIP calls Route Call to PSTN Unconditional Call Forward to PSTN Unconditional Call Forward to VoIP AR Innovative IP Voice amp Video 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 For example if 64 is configured there will be at least 64kbps reserved for VolP The primary function of this setting is to reserve bandwidth for VolP 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 For example if 128 is configured there will be at least 128kbps reserved for VoIP The primary function of this setting is to reserve bandwidth for VoIP When set to Yes the HT503 acts as an UPnP gateway for your UPnP enabled applications UPnP Unive
6. The configuration files can be downloaded via TFTP or HTTP from the central server A service provider or an enterprise with large deployment of HT503 can easily manage the configuration and service provisioning of individual devices remotely from a central server Grandstream provides a licensed provisioning system called GAPS that can be used to support automated configuration of HT503 GAPS Grandstream Automated Provisioning System uses enhanced NAT friendly TFTP or HTTP thus no NAT issues and other communication protocols to communicate with each individual HT503 for firmware upgrade remote reboot etc Grandstream provide GAPS Grandstream Automated Provisioning System service to VolP service providers It could be either simple redirection or with certain special provisioning settings Initially upon booting up Grandstream devices by default point to Grandstream provisioning server GAPS based on the unique MAC address of each device GAPS provision the devices with redirection settings so that they will be redirected to customer s TFTP or HTTP 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 G
7. 0 0 Syntax f1 freq vol f2 freq vol c on1 off1 on2 off2 on3 off3 Note freq 0 4000HZz 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 Select the ATA to report the log level Default is NONE The level is either one of DEBUG INFO WARNING or ERROR Syslog messages a
8. 0 0 9 Last Updated 9 2007 HARDWARE SPECIFICATION ndstream Innovative IP Voice amp Video The table below lists the hardware specification of HT503 TABLE 3 HT503 HARDWARE SPECIFICATION LAN interface WAN interface FXS telephone port FXO telephone port PSTN Port LED Universal Switching Power Adaptor Dimension Weight Temperature Humidity Compliance 1xRJ45 10 100 Mbps Port 1xRJ45 10 100 Mbps Port 1 x FXS RJ11 1x PSTN pass through and life line port Power WAN LAN PHONE and LINE Green Input 100 240 VAC 50 60 Hz Output 12VDC 0 5A UL certified 25mm x 115mm x 75mm when laying flat 115mm x 25mm x 75mm standing up Approximately 0 6lbs 0 3kg Operational 32 104 F or 5 45 C Storage 10 130 F 10 90 non condensing FE CE Grandstream Networks Inc HT503 User Manual Page 8 of 37 Firmware 1 0 0 9 Last Updated 9 2007 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 TABLE 4 HT503 IVR MENU DEFINITIONS Men Voice Prompt Main Menu 01 02 03 04 05 07 12 13 14 15 16 17 47 99 Enter a Menu Option DHCP Mode Static IP Mode IP Address IP address Subnet IP addres
9. a 12 digit IP address to make a direct IP call after dial tone See Make a Direct IP Call Press 9 to reboot the device or Enter encoded MAC address to restore factory default setting See Restoring Factory Settings Automatically returns to main menu Grandstream Networks Inc HT503 User Manual Page 9 of 37 Firmware 1 0 0 9 Last Updated 9 2007 ndstream Innovative IP Voice amp Video NOTE e shifts down to the next menu option e returns to the main menu e 9 functions as the ENTER key in many cases to confirm an option e 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 e Key entry can not be deleted but the phone may prompt error once it is detected PLACING PHONE CALLS CALLING PHONE OR EXTENSION NUMBERS There are currently two methods to make an extension number call a Dial the numbers directly and wait for 4 default seconds b Dial the numbers directly and press assuming that use as dial key is selected in the web configuration EXAMPLES e 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 sec
10. 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 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 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 2 digits number exclude 8 5 any digit of 3 4 or 5 147 any digit 1 4 or 7 f lt 2 011 gt replace digit 2 with 011 when dialing e Example 1 369 11 1617xxxxxxx Allow 311 611 911 and any 10 digit numbers of leading digits 1617 e Example 2 1900x lt 1617 gt xxxxxxx Block any number of leading digits 1900 and add prefix 1617 for any dialed 7 digit numbers e Example 3 1xxx 2 9 xxxxxx lt 2 011 gt x Allow any length of number with leading digit 2 and 10 digit numbers of leading digit 1 and le
11. for G726 16 Default value is 98 Range is from 96 to 127 Defines payload type for G726 24 Default value is 99 Range is from 96 to 127 Defines payload type for G726 40 Default value is 103 Range is from 96 to 127 Defines payload type for G729E Default value is 102 Range is from 96 to 127 Default is No VAD allows detecting the absence of audio and conserves bandwidth by preventing the transmission of silent packets over the network 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 T 38 Auto Detect FolP by default or fax Pass Through must use PCMU PCMA Default is Callee This decides whether Caller or Callee sends out the re invite for T 38 or Fax Pass Through Select either Fixed or Adaptive based on network conditions Select Low Medium or High based on network conditions Secure RTP protocol used for media transmission over VoIP Disabled by default Other modes are enabled but not forced amp enabled and forced Dependent on standard phone type and location Bellcore Telcordia ETSI FSK ETSI DTMF SIN 227 BT amp NTT Japan A value of level for Caller ID information sent by a FXS port to phone connected to it 40 OdB Default 20dB If set to Yes polarity will be reversed upon call establishment and termination Default is No Set it to Yes of the
12. headers in outgoing INVITE messages 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 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
13. port To receive PSTN calls pick up the phone when it rings To complete a PSTN call press the PSTN access code 00 is default or any number configured in the web configuration to switch to the PSTN line listen for a dial tone then dial the number It the HT503 loses power it will function as a jack enabling a direct connection to the PSTN Line If the 503 loses power or lost registration with SIP server device will switch to mode when PSTN line will be transparently connected directly to phone connected to FXS port It will function as a jack enabling a direct connection to the PSTN Line 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 Note 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 The special continuous tone is played if the pin code is configured otherwise the caller will hear a dial tone Enter the pin code configured on the configuration page The caller will hear a dial tone and be connected to the PSTN line if the pin code is valid If the pin code is invalid the continuous tone is played to prompt caller to enter the pin code again The user may try up to 3 times to enter a correct pin code After t
14. same LAN segment 3 Please go to File gt Configure gt 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 phone s web configuration page 5 Configure the Firmware Server Path with the IP address of the PC 6 Update the change and reboot the unit The end user can also choose to download the free HTTP server from http httpd apache org or use Microsoft IIS web server Grandstream Networks Inc HT503 User Manual Page 35 of 37 Firmware 1 0 0 9 Last Updated 9 2007 ndstream Innovative IP Voice amp Video 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 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 wil
15. the FXS port and a web based 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 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 internet disconnect survivable fail over to PSTN support dual 10 100Mbps Ethernet ports with integrated high performance NAT router a flexible dial plan and a broad range of popular voice codecs TABLE 2 HT503 TECHNICAL SPECIFICATIONS Interfaces 1 FXS telephone port RJ11 1 FXO PSTN line port RJ11 with lifeline support e TWO 2 10M 100 Mbps ports RJ45 with integrated Nat router eeano Protocol Support TCP UDP IP RTP RTCP HTTP HTTPS ARP RARP ICMP DNS DHCP NTP TFTP PPPoE STUN amp TELNET protocols LED Indicators Power WAN LAN PHONE and LINE _ RESET Button Factory Reset Button J yy Device Management Web interface or via secure AES encrypted central configuration file for mass deployment Support device configuration via built in IVR Web browser or central configuration file through TFTP or HTTP Support Layer 2 802 1Q VLAN 802 1p and Layer 3 QoS ToS DiffS
16. the HT503 will attempt to retrieve the new image files by downloading them into the SRAM During this stage the HT503 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 fails for any reason e g TFTP HTTP 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 process and simply boot using the existing code image in the flash 4 Firmware upgrades usually take around 2 minutes when 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 Please check the Services section of Grandstream s Web site to obtain our public TFTP server s IP address 5 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 Directions to download a free TFTP Server 1 Unzip the file and put all of them under the root directory of the TFTP server 2 Putthe PC running the TFTP server and the HT503 device in the
17. user to access the Web Configuration Menu User can put new password here This field is case sensitive with maximum of 25 characters This is the device s internal HTTP server port Default is 80 Default is set to YES Telnet access is allowed to the device in this case Used only for special purposes such as debugging and troubleshooting List of available commands will be shown by pressing gt help command from telnet console e 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 The HT503 will acquire its IP address from DHCP in the network e PPPoE settings are usually for DSL ADSL modem users The HT503 will attempt to establish a PPPoE session if PPPoE account is set e 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 This option specifies the name of the client This field is optional but may be required by some Internet Service Providers Default is blank This option specifies the domain name that client should use when resolving hostnames via the Domain Name System Default is blank This option is used by clients and servers to exchange vendor specific information Default is blank PPPoE username Necessary if your ISP requires you to use a PPPoE Point to Point Protocol over Ethernet connection PPPoE account passwo
18. 0 9 Last Updated 9 2007 ndstream Innovative IP Voice amp Video FIGURE 3 SCREENSHOT OF CONFIGURATION LOG IN PAGE Grandstream Device Configuration Password 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 It is most likely the VoIP service provider has provisioned the device and configured for you therefore the password has already been changed After a correct password is entered in the login screen the embedded web server will respond with the Configuration pages which are explained in details below TABLE 7 HT503 DEVICE STATUS PAGE DEFINITIONS MAC Address WAN IP Address Product Model Software Version System Uptime PPPoE Link Up NAT Port Status 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
19. 7 of 37 Last Updated 9 2007 AR Innovative IP Voice amp Video TABLE 11 HT503 FXO PORT SETTINGS PAGES DEFINITIONS Account Active SIP Server Outbound Proxy SIP Transport NAT Traversal STUN SIP User ID Authenticate ID Authenticate Password Name Use DNS SRV User ID is Phone Number SIP Registration Unregister on Reboot Outgoing Call Without Registration Register Expiration Local SIP Port Local RTP Port Use Random Port Refer to Use Target Contact Validate incoming message SIP T1 Timeout When set to Yes the FXO port is activated SIP Server s IP address or Domain name provided by VoIP Service Provider IP address or Domain name of Outbound Proxy or Media Gateway or Session Border Controller Used by HT503 for firewall or NAT penetration in different network environments If symmetric NAT is detected STUN will not work and ONLY way to correct the problem is to use the outbound proxy User can select UDP TCP or TLS 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 I
20. 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 Note In order to receive the caller ID information the delay should be set to a value 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 Otherwise the dial plan thinks it is the end of the digit input Wait for Dial Tone Wait for Dial tone is used for one stage VoIP to PSTN calls If set to Yes the device will first obtain a PSTN line and a dial tone from a central office After obtaining the dial tone the digits dialed will be sent to the central office Stage Method 1 2 This configuration is applicable for VoIP to PSTN calls and indicates one or two stage dialing methods Note General settings have the same meaning as explained in the FXS page definitions Here they are described only as parameters related to the FXO port SAVING THE CONFIGURATION CHANGES Once a ch
21. Default is Yes 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 Grandstream Networks Inc HT503 User Manual Firmware 1 0 0 9 Page 22 of 37 Last Updated 9 2007 System Ring Cadence Call Progress Tones Lock Keypad Update Disable Voice Prompt Disable Direct IP Calling Life Line Mode NTP server Syslog Server Syslog Level Download Device Configuration AR Innovative IP Voice amp Video Configuration option for FXS port ring cadence for all incoming calls Syntax c on1 off1 on2 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 11 350 13 f2 440 13 c
22. E E EERE AR EEEE EEE annann annann 14 Forward CalletoboG hN 15 Forward Calletovolb tnt nn rannen unnn nanan EAEE EEan EAEE E EEEE nnan aa nnan nannan 15 One Stade Didin DEE 15 Fax SUDI s EEN 15 GALE FEATURES dee 16 LED Light Pattern Indieatton lnsanpa taon tann O ara k A KRAKE R RR RRR RR 16 CONFIGURATION GUD E ssclcsva lla Oa alara EE 17 Configuring HT503 through Voice Prompt ssssssssssrssrrssrssrrssrrssrresirssrisssrnssrnsstnnstnnsrnnntnnnnnnnnnnnnnn nnno 17 DACP MOJE ee a gt eo PS Ste Pe 2 SR OP OO SOT E a 17 e Ile 17 TET P Semer Addres S aoa a asa a aa a s a aa 17 Firmware Server IP Address 17 Configuration Server IP Address asasi assavedusondtavdka t skask jastakisat alda all HVAD Kan d HN K AS kka akak aaa 17 Upgrade ges e 17 Firmware Upgrade Mode ssssseneeeeesenssenssenssensstnsstnsstensstnsstnsstnsstnssttnstnnstenstenstnnsnnnttnnntnnnstentnnnnnnnennnet 17 Configuring HT503 with Web Browser aaiiaaaaaasaaasaaassaassaassaanaaaasaaanaaanaaanaaanaaansaanaaannananaaanaaanaaanaannna 18 Access the Web Configuration Mem 18 End User Configuration EE 18 Advanced Configuration and FXS FXO Ports Parameierg 22 Saving the Configuration Changes ssssesseesisesirssirssrnssinssinssinssttnsttnstnstnsrunstnnstnastnnsnnnnnnnnn annann nn 32 Remote Reboot of the HT503 aiaaaaaaaaaaaaaaasaaaaanaasananaanannusannananannanannanannnnnnnnnananannnnnnnnnnnnnnnnanananunana 32 Configuration through a Ce
23. Ostream Innovative IP Voice amp Video Grandstream Networks Inc HT503 FXS FXO Port Analog Telephone Adaptor HT503 User Manual www grandstream com Firmware Version 1 0 0 9 support grandstream com ndstream Innovative IP Voice amp Video TABLE OF CONTENTS Firmware 1 0 0 9 HT503 USER MANUAL WELCOME inna a a aad arena 4 Safety Compliances aniiiaaaai aasaaaaasaaaaannaanannnnnnannnnnnnnnnnannnnnannnnnannnnnannnnnannnnnannnnnnannnnnannnnnannnnannnnnannnananana 4 NEITS Ee eege I EREET ee A INSTALLATION Gia a a a eE 5 Eguipmen PACKAGING teiiissieecesaaedeccesavencaravecsrtaxansdenstanssivazaecanhechansdtasaashacegaarsanecdevnilavassdnusaanciieasneinaannnes 5 Gei ele YOURA TA EE 5 Five easy steps to install the HTbO cc cceccscecsesseeeceeeeeeeseeeaeeeseseeaeeeceesaeeeseesaeeessesaeeesecaaeesseseeeessaaes 6 leien ge TEE 7 Hardware Gpechication isi KAK AKAU KIA NNO KRK N RAAR RIKERA K K ARR K NN KRL ARAR KERA ARRENA RENA KARAER REKRAI 8 BASIC elle TER 9 Get Familiar with Voice Prompt A 9 Placing Vi le Ier EE 10 Calling Phone or Extension Numbers 10 Direct ak Get EE 10 CU lee EE 11 EEEa A E asta arr karra 11 UR dE TEE 11 SB eu le sreski ont ensbeeedenawennebteaneeanadeernaaceneiaseee 12 PS TIN Pass Re EE 13 Vie Ne ERC Rer EE 13 PSTN to VoIP Callls 1170 EE A E A AS EEE EEE dtaa urkuna abu skaa ab sdana t usai a akvanknsasudun 14 Route CalletoboG hN rn nenn nn unnn kanaa EAEE EAEan EAE
24. RFC2833 or SIP Info for DTMF digit transmission e The special continuous tone is the prompt to enter a valid PIN code If a caller doesn t 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 e 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
25. RMWARE 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 and HTTP FIRMWARE UPGRADE MODE Select voice menu option 17 to choose firmware upgrade mode There are three options 1 always check 2 check when pre suffix changes and 3 never upgrade Grandstream Networks Inc HT503 User Manual Page 17 of 37 Firmware 1 0 0 9 Last Updated 9 2007 ndstream Innovative IP Voice amp Video CONFIGURING HT503 wu 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 e 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 addre
26. Updated 9 2007 ndstream Innovative IP Voice amp Video 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 Itis 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 VolP 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 resources html SAFETY COMPLIANCES The HT503 adaptor complies with FCC CE and various safety standards The HT503 power adaptor is compliant with UL
27. 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 Enable Call Waiting for all subsequent calls Block Caller ID per call Dial 67 number No dial tone is played in the middle Send Caller ID per call Dial 82 number No dial tone is played in the middle Call Return Service Dial 69 and the phone will dial the last incoming phone number received Disable Call Waiting per call Dial 70 number No dial tone is played in the middle Enable Call Waiting per call Dial 71 number No dial tone is played in the middle Unconditional Call Forward Dial 72 and then the forwarding number followed by Wait for dial tone and hang up dial tone indicates successful forward Cancel Unconditional Call Forward To cancel Unconditional Call Forward dial 73 wait for dial tone then hang up Enable Do Not Disturb DND When enabled all incoming calls are rejected Disable Do Not Disturb DND When disabled incoming calls are accepted Blind Transfer Busy Call Forward Dial 90 and then the forwarding number followed by Wait for dial tone then hang up Cancel Busy Call Forward To cancel Busy Call Forward dial 91 wait for dial tone then hang up Delayed Call Forward Dial 92 and then the forwarding number followe
28. ading exchange number between 2 and 9 If leading digit is 2 replace leading digit 2 with 011 before dialing 3 Default Outgoing x Default is No When set to Yes a SUBSCRIBE for Message Waiting Indication will be PAP Oy Grandstream Networks Inc HT503 User Manual Firmware 1 0 0 9 Page 29 of 37 Last Updated 9 2007 Send Anonymous Anonymous Call Rejection Special Features Session Expiration Min SE Caller Request Timer Callee Request Timer Force Timer UAC Specify Refresher UAS Specify Refresher Force INVITE Preferred Vocoder G723 Rate iLBC Frame Size iLBC Payload Type G726 16 Payload G726 24 Payload Type G726 40 Payload Type G729E Payload Type VAD Symmetric RTP Fax Mode Dag Innovative IP Voice amp Video sent periodically When set to Yes the From header will have value anonymous blocking Caller ID Pre configured account user Id is put in the P Asserted Identity field 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
29. am Innovative IP Voice amp Video EXAMPLES 1 Ifthe 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 NOTE When placing a direct IP call the Use Random Port should be set to NO 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 CALL WAITING This function is applicable on FXS port for VoIP calls only If the call waiting feature is enabled the user will hear a special stutter tone if there is another call on the line Press the flash button to place the current party on hold and switch to the other call Pressing the flash button toggles between two active calls The HT503 also provides CWCID call
30. ange 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 Remote Reboot of the HT503 The administrator can remotely reboot the HT503 by clicking on the Reboot button at the bottom of the configuration page Once done the following screen will be displayed to indicate that rebooting is underway You can login again after about 30 seconds Grandstream Networks Inc HT503 User Manual Page 32 of 37 Firmware 1 0 0 9 Last Updated 9 2007 ustream Innovative IP Voice amp Video FIGURE 4 SCREENSHOT OF REBOOTING SCREEN The device is rebooting now You may relogin by clicking on the link below in 30 seconds Click to relogin NOTE Interrupting the booting up process could permanently damage the device Grandstream Networks Inc HT503 User Manual Page 33 of 37 Firmware 1 0 0 9 Last Updated 9 2007 ndstream Innovative IP Voice amp Video 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 MAC address of the HT503
31. are 1 0 0 9 Page 25 of 37 Last Updated 9 2007 Subscribe for MWI Send Anonymous Anonymous Call Rejection Special Features Session Expiration Min SE Caller Request Timer Callee Request Timer Force Timer UAC Specify Refresher UAS Specify Refresher Force INVITE Preferred Vocoder AR Innovative IP Voice amp Video e Example 3 1xxx 2 9 xxxxxx lt 2 011 gt 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 3 Default Outgoing x Example of a simple dial plan used in a Home Office in the US 1900x lt 1617 gt 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 e 41900x prevents dialing any number started with 1900 lt 1617 gt 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 e 011 2 9 x allows international calls starting with 011 e 3469 11 allow dialing special and emergency numbers 311 411 611 and 911 Default is No When set to Yes a SUBSCRIBE for Message Waiting Indication will be sent periodically When set to Yes the From header along with Privacy and P_Asserted_Identity
32. ash 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 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 16688 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 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 upgrades and provisioning In check for new firmware every field enter the number of days to set the frequency in which the HT503 will check the server for firmware or configuration upgrades When set to No the HT503 will only upgrade at boot time You can also have the HT503 check for updates only when the firmware file prefix suffix changes The
33. be for MWI AR Innovative IP Voice amp Video Maximum retransmission interval for non INVITE requests and INVITE responses Sends DTMF using RFC2833 Sends DTMF as inband in audio Sends DTMF via RTP according the RFC2833 Send DTMF as a SIP INFO message Default is No If set to Yes flash will be sent as DTMF event Default is Yes Toggles support for advanced call features and star code functions 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 Turns off the Call Waiting feature Default is No Default is No This is to disable the stutter Call Waiting Tone during a CWC The CWCID will still be displayed 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
34. ccsesesececeeececeesseaeseeeescucseseeaececceseeccesaeaeceeeeesescseaaeeeeeeseeeses 9 TABLE 5 HT503 CALL FEATURE DEFINITIONS cccccesscseceeececceseseeeeceeececseseaaaeaecesececeeseaseaeceeecesseseaaeseeeeseeenes 16 TABLE 6 HT503 LED DEFINITIONS 2112 2224 dia an l s sasbdds a d d us a ssvd stanbagdal nd d tssi aka stundasss ndka adaai s dasasavdann inar a 16 TABLE 7 HT503 DEVICE STATUS PAGE DEEINTTIONS 19 TABLE 8 HT503 BASIC SETTINGS PAGE DEFINITIONS ssssssininsenseininrnsersirinnnnnssensrnirnnnnsensrinnnnnsennrernnns 20 TABLE 9 HT503 ADVANCED SETTINGS PAGE DEFINITIONS u aaaaiiaaaaaaaaaaasaaaaaasasaanaaaananasaaaaaaaaaanaauaaaaaaaaaaaaaaaaaa 22 TABLE 10 HT503 FXS PORT SETTINGS PAGES DEFINITIONS auiiaaaaaaaaaaasaaaaaasasnananaaaanasaaaaaaaaanaaaaaaaaaaaaaaaaaaaaaa 24 TABLE 11 HT503 FXO PORT SETTINGS PAGES DEEINTIONS 28 TABLE OF GUI INTERFACES HT503 USER MANUAL http www grandstream com user_manuals GUI GUI HT503 rar 1 SCREENSHOT OF CONFIGURATION LOGIN PAGE 2 STATUS CONFIGURATION PAGE DEFINITIONS 3 SCREENSHOT OF BASIC SETTINGS CONFIGURATION PAGE 4 SCREENSHOT OF ADVANCED SETTINGS CONFIGURATION PAGE 5 SCREENSHOT OF FXS ACCOUNT CONFIGURATION 6 SCREENSHOT OF FXO ACCOUNT CONFIGURATION 7 SCREENSHOT OF CALL PROGRESS TONES CONFIGURATION PAGE 8 SCREENSHOT OF SAVED CONFIGURATION CHANGES 9 SCREENSHOT OF REBOOT PAGE Grandstream Networks Inc HT503 User Manual Page 3 of 37 Firmware 1 0 0 9 Last
35. d by Wait for dial tone then hang up Cancel Delayed Call Forward To cancel Delayed Call Forward dial 93 wait for dial tone then hang up Toggles between active call and incoming call call waiting tone If not in conversation flash nook will switch to a new channel for a new call Pressing pound sign will serve as Re Dial key LED Light Pattern Indication TABLE 6 HT503 LED DEFINITIONS a 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 port 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 Grandstream Networks Inc HT503 User Manual Page 16 of 37 Firmware 1 0 0 9 Last Updated 9 2007 ndstream Innovative IP Voice amp Video 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 FI
36. e 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 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 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 e Users can choose whether or not to apply password protection for VolP 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 e there is no authentication required for callers on the use of PSTN line through HT503 e When a PIN is configured for VOIP to PSTN call flow the VoIP device that calls into the HT503 FXO account needs to configure
37. e URL string or the IP address and port if different from 5060 of the SIP proxy server e g the following are some valid examples sip my voip provider com or sip my company sip server com or 192 168 1 200 5066 IP address or Domain name of Outbound Proxy or Media Gateway or Session Border Controller Used by ATA for firewall or NAT penetration in different network environment If symmetric NAT is detected STUN will not work and ONLY Outbound Proxy will work User can select UDP or TCP or TLS This setting decides whether the NAT traversal mechanism is activated It should be set to Yes if the device is behind a NAT router If no outbound proxy is configured a STUN server needs to be set to activate STUN detection mechanism Usually ITSP will 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 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 Do NOT include the preceding sip scheme or the host portion of the SIP address in this field ID used for authentication usually same as SIP user ID but could be different and decided by ITSP Password for ATA to regis
38. e connected to the PSTN line Grandstream Networks Inc HT503 User Manual Page 5 of 37 Firmware 1 0 0 9 Last Updated 9 2007 ndstream Innovative IP Voice amp Video 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 Insert the Ethernet cable into the WAN port of HT503 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 HT503 if it is being used as a router 5 Insert the power adapter into the HT503 and connect it to a wall outlet FIGURE 2 INTERCONNECTION DIAGRAM OF THE HT503 H Er HK j Vib Internet ADSL Cable Modem Ethernet TT WAN Analog Phone Cordless Grandstream Networks Inc HT503 User Manual Page 6 of 37 Firmware 1 0 0 9 Last Updated 9 2007 ndstream Innovative IP Voice amp Video 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
39. e will pick randomly generated SIP and RTP ports This is usually necessary when multiple HandyTone ATAs are behind the same NAT Default is No If set to Yes then for Attended Transfer the Refer To header uses the transferred target s Contact header information 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 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 Grandstream Networks Inc HT503 User Manual Firmware 1 0 0 9 Page 24 of 37 Last Updated 9 2007 SIP T2 Interval 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 Disable Visual MWI Ring Timeout No Key Entry Timeout Early Dial Dial Plan Prefix Use as Dial key Dial Plan AR Innovative IP Voice amp Video Maximum retransmission interval for non INVITE requests and INVITE responses This parameter 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 Default is No If set to yes flash will be sent as DTMF event Default is Yes Toggl
40. ed in back panel of the device for approximately 8 seconds All port LEDs will turn on and device will restart itself NOTE 1 Factory Reset will be disabled if the Lock keypad update is set to Yes 2 Please be aware by default the HT503 WAN side HTTP access is disabled After a factory reset the device s web configuration page can be accessed only from its LAN port 3 If the HT503 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 Grandstream Networks Inc HT503 User Manual Page 37 of 37 Firmware 1 0 0 9 Last Updated 9 2007
41. er Path Config Server Path Firmware File Prefix Firmware File Postfix Config File Prefix Config File Postfix Automatic Upgrade Authenticate Conf File Firmware Key SSL Certificate SSL Private Key SSL Private Key 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 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 Fl
42. erv MPLS Auto manual provisioning system NAT friendly remote software upgrade via TFTP HTTP for deployed devices including behind firewall NAT Syslog support DHCP Server Client Yes Audio Features Advanced Digital Signal Processing DSP Dynamic negotiation of codec and voice payload length Support for G 723 1A G 729A B E G 711 G 726 40 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 amp loss concealment PLC amp G 168 compliant Line Echo Cancellation Support volume amplification Support configurable Call Progress Tones a Call Handling Features Caller ID display or block Call waiting caller ID Call waiting flash Call transfer hold call forward do not disturb 3 way conferencing M Network and Manual or dynamic host configuration protocol DHCP network setup RTP and NAT Provisioning support traversal via STUN Fax over IP T 38 compliant Group 3 Fax Relay up to 14 4kpbs and auto switch to G 711 for Fax EE through pending Fax Data pump V 17 V 19 V 27ter V 29 for T 38 fax relay Security DIGEST authentication and encryption using MDS and MD5 sess nan Physical Design Stylish and compact design small universal power supply ideal for travel Grandstream Networks Inc HT503 User Manual Page 7 of 37 Firmware 1
43. es support for advanced call features and star code functions This parameter allows users to configure a User ID or extension number to be automatically dialed when offhook Please note that only the user part of a SIP address needs to be entered here The HT503 will automatically append the and the host portion of the corresponding SIP address Note User will need this IP address when accessing the IVR via the web configuration page 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 Customizes Ring Tones 1 to 3 with an associated Caller ID When selected the device will ONLY use this ring tone when the incoming call is set up for Caller ID The System Ring Tone is used for all other calls When selected with no Caller ID configured the selected ring tone will be used for all incoming calls Default is No 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 Sets the time in which an incoming call will stop ringing when not picked up 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
44. f the NAT Traversal field is set to Yes with no specified STUN server the HT503 will periodically every 20 seconds or so send a blank UDP packet with no payload data to the SIP server to keep the hole on the NAT open User account information provided by VoIP service provider ITSP Usually in the form of digit similar to phone number or actually a phone number The SIP service subscriber s ID used for authentication Can be identical to or different from SIP User ID SIP service subscriber s account password SIP service subscriber s name for Caller ID display Default is No If set to Yes the client will use DNS SRV to look up the server 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 Controls whether the HT503 needs to send REGISTER messages to the proxy server The default setting is Yes Default is No If set to Yes the SIP user s 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 HT503 refreshes its registration with the specified registrar The default interval is 60 minutes or 1 hour The max
45. hree 3 tries the HT503 hangs up After the caller hears a dial tone from PSTN line the caller can place the next call The user can hit the key to identify the end of the pin code or wait 4 seconds for a new dial tone and then dialing the PSTN number 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 l 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 doesn t 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 Grandstream Networks Inc HT503 User Manual Page 13 of 37 Firmware 1 0 0 9 Last Updated 9 2007 ndstream Innovative IP Voice amp Video e On the web configuration page if the Forward to PSTN 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 PSTN To VoIP CALLS This function is available using the FXO port Th
46. imum interval is 65535 minutes about 45 days Defines the local SIP port the HT503 will listen and transmit The default value for FXS port is 5062 This parameter defines the local RTP RTCP port pair used by the HandyTone ATA It is the base RTP port for FXO channel 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 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 target s contact header information 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 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 Grandstream Networks Inc HT503 User Manual Firmware 1 0 0 9 Page 28 of 37 Last Updated 9 2007 SIP T2 Interval DTMF Payload Type DTMF in Audio DTMF via RFC2833 DTMF via SIP INFO Send Flash Event Enable Call Features Proxy Require Use NAT IP Disable Call Waiting Disable Call Waiting Tone Ring Timeout Early Dial Dial Plan Prefix Use as Dial Key Dian Plan Subscri
47. k Flash for old model phones to get a dial tone A dials 23 then C s number then or wait for 4 seconds If C answers the call then A presses FLASH to bring B C in the conference If C does not answer the call A can press FLASH back to talk to B If A presses FLASH during conference C will be dropped out ME Er e Note Enable Call Feature has to be set to YES in FXS PORT in the web configuration page Bellcore Style 3 way Conference To use the Bellcore Style conference the Use Bell style 3 way Conference field in FXS PORT web configuration must be enabled Assume that parties A and B are in conversation Party A using the HT503 wants to bring C into a 3 way conference Grandstream Networks Inc HT503 User Manual Page 12 of 37 Firmware 1 0 0 9 Last Updated 9 2007 ndstream Innovative IP Voice amp Video A presses FLASH on the analog phone or Hook Flash for old model phones to get a dial tone A dials C s number then or wait for 4 seconds If C answers the call then A presses FLASH to bring B C in the conference If C does not answer the call A can press FLASH back to talk to B If A presses FLASH during the conference C will be dropped out ech BR N Note Party A is the call initiator for both calls with party B and party C PSTN PASS THROUGH HT503 supports PSTN pass through using the FXS port The user can place and receive PSTN calls using analog phone connected to FXS
48. l 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 An outbound call will be sent directly to the assigned FXO port account where there the HT503 will initiate a call to the local CO The RequestURI header in the INVITE message contains the phone number used to initiate the call to the local CO FAX SUPPORT HT503 supports FAX in two modes 1 T 38 Fax over IP and 2 fax pass through T 38 is the preferred method because it is more reliable and works well in most network conditions If the service provider supports T 38 please use this method by selecting Fax mode to be T 38 default If the service provider does not support T 38 pass through mode may be used To send or receive faxes in fax pass through mode users must select all the Preferred Codecs to be PCMU PCMA G 711 p a Grandstream Networks Inc HT503 User Manual Page 15 of 37 Firmware 1 0 0 9 Last Updated 9 2007 GH Innovative IP Voice amp Video CALL FEATURES TABLE 5 HT503 CALL FEATURE DEFINITIONS Key 30 31 47 50 51 67 82 69 70 71 72 73 78 79 87 90 91 92 93 Flash Hook Call Features Block Caller ID for all subsequent calls Send Caller ID for all subsequent calls Direct IP Calling Dial 47 IP
49. l issue request for configuration file named CfQXXXXXXXXXXXX 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 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 to Yes the Service Provider can use P193 Auto Check Interval in minutes default and minimum is 60 minutes to have the devices periodically check with either Firmware Server or Config Server however they are defined This allows the device to 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 the Server Provider can spread the Firmware or Configuration File download in min
50. ntral Geer 34 la 35 Firmware Upgrade through TFTP HTTP ssssssssssesssesssesssesssssesrssnnssnustnrnstnuttnntnnustnustnutnnsuntunnnnnnnnnennnnnn 35 Configuration File Download 36 Firmware and Configuration File Prefix and Postfix aaaaassaassaassaassaassaassaannaansnannaannaannannnaannannnnannna 36 Managing Firmware and Configuration File Download aaaaiaaaaaaaaaaaaaaasaaaaaasaanaannannanaannannaananaannaanina 36 RESTORE FACTORY DEFAULT SETTING va a a al laa kal mak aaa nalva a ENTREE NEEN 37 Grandstream Networks Inc HT503 User Manual Page 2 of 37 Last Updated 9 2007 TABLE OF FIGURES ndstream Innovative IP Voice amp Video HT503 USER MANUAL FIGURE 1 CONNECTING THE HT503 aaaaaiaaaaaiaaaaaaaasasaaaaanaaaanasnaaanaaaanaanananaananuaaananaanannadaaaanaaaanasaaaanaananaaaaaaaaaa 5 FIGURE 2 INTERCONNECTION DIAGRAM OF THE HT503 i naaaaiaaaaaaaaasaasaaaaaaasaaaaaaaaaananuaaananaananudaaaaaaananaadaaaaaaaa 6 FIGURE 3 SCREENSHOT OF CONFIGURATION LOG NPBAGE 19 FIGURE 4 SCREENSHOT OF REBOOTING SCHEEN 33 TABLE OF TABLES HT503 USER MANUAL TABLE 1 DEFINITIONS OF THE HT503 CONNECTORS 4uuuaisaaaaanaaaasaaaaannaaaannanaannaanannanaannaaaannanaannanaannanaannanaanaaaaaanna 5 TABLE 2 HT503 TECHNICAL SPECIFICATIONS aaaiiaaaaaaaasaaaaaaaaanasnananaaaanannananaanananaananaananunaanaanananuaaanaaanananaananaaaaa 7 TABLE 3 HT503 HARDWARE GPECIEIGATION 8 TABLE 4 HT503 IVR MENU DEFINITIONS c ccccccc
51. number is always used for firmware upgrade Current release is 1 0 0 9 Bootloader current version is 1 0 0 7 Core current version 1 0 0 18 Base current version is 1 0 0 43 This shows system up time since last reboot This shows whether the PPPoE is up if connected to DSL modem This shows what kind 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 Busy Delayed Forward Forward FXS On Hook Registered Yes 613 FXO Idle Registered No 614 e Both FXS port and FXO port are registered with this SIP Server e FXS Port user has set Do Not Disturb e FXS Port user has set his calls to be forwarded unconditionally to ext 613 FXO Port user has set his calls to forward to 614 when his phone is busy Grandstream Networks Inc HT503 User Manual Page 19 of 37 Firmware 1 0 0 9 Last Updated 9 2007 E son Innovative IP Voice amp Video TABLE 8 HT503 BASIC SETTINGS PAGE DEFINITIONS End User Password Web Port Telnet Server IP Address DHCP hostname DHCP domain DHCP vendor class ID PPPoE account ID PPPoE password PPPoE Service name Preferred DNS Time Zone Self Defined Time Zone Language Device Mode NAT Maximum Ports This contains the password for end
52. o PSTN volume Default OdB 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 If voice volume is too low at the other end PSTN side user may increase the far end volume using the Tx Gain Level parameter under the FXO Port Configuration page Default is Yes This value should be used in case the PSTN provider uses line power drop to indicate call completion to the end point In this case the HT503 will search for a power drop for a preconfigured time frame to disconnect such calls from a VoIP extension This is a preconfigured value of duration for a line power drop used by specific service providers For example for a configured value of 500ms the device will ignore any random voltage drops on the line if duration of such drop is less than 500ms and the call will NOT be considered as terminated This is useful to prevent unnecessary call drops in some low quality PSTN lines If set to Yes arrived Busy Tone is used as the disconnect signal In certain countries the central office will send a special busy tone to indicate when a call is disconnec
53. onds e 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 HT with an analog phone and another VoIP Device to talk to each other in an ad hoc fashion without a SIP proxy This kind of VoIP calls can be made between two parties if e Both HT503 and other VoIP Device i e another Handytone ATA or Budgetone SIP phone or other VoIP unit have public IP addresses or e Both HT503 and other VoIP Device are on the same LAN using private IP addresses or e Both HT503 and other VoIP Device can be connected through a router using public or private IP addresses with necessary port forwarding or DMZ TO PLACE A DIRECT IP CALL 1 Pick up the analog phone or use the speakerphone 2 Access the voice menu prompt by dial 3 Dial 47 to access the direct IP call menu 4 At voice prompt Direct IP Calling and dial tone enter a 12 digit target IP address to make a call Destination ports can be specified by using 4 encoding for followed by the port number Grandstream Networks Inc HT503 User Manual Page 10 of 37 Firmware 1 0 0 9 Last Updated 9 2007 ndstre
54. randstream Networks Inc HT503 User Manual Page 34 of 37 Firmware 1 0 0 9 Last Updated 9 2007 ndstream Innovative IP Voice amp Video 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 firmware mycompany com 6688 Grandstream 1 0 0 9 e g 168 75 215 190 NOTES 1 The TFTP server in IP address format can be configured via IVR Please refer to the CONFIGURATION GUIDE section for instructions If the TFTP server is in FQDN format it must be set via the web configuration interface 2 End users recommended using our TFTP server lts address can be found at http www grandstream com firmware html Currently the TFTP server your HT503 can be upgraded from has an IP address 168 75 215 189 For companies we recommend to maintain their own TFTP HTTP server for upgrade and provisioning procedures 3 Once a Firmware Server Path is set the user needs to update the settings and reboot the device If the configured firmware server is found and a new code image is available
55. rd This field is optional If your ISP uses a service name for the PPPoE connection enter the service name here Default is blank The address of your preferred DNS server This parameter controls how the displayed date time will be adjusted 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 This indicates a time zone with 6 hours offset with 1 hour ahead which is U S central time If 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 The 1 number indicates Month 1 2 3 12 for Jan Feb Dec The 277 number indicates the nth iteration of the weekday ae Sunday 314 Tuesday The 3 number indicates weekday 0 1 2 6 for Sun Mon Tues Sat Therefore this example is the DST which starts from the second Sunday of March to the 1 Sunday of November Languages supported with the voice prompt This parameter controls whether the device is working in NAT router mode or Bridge mode Save the setting and reboot prior to configuring the HT503 The number of ports that can be managed while in NAT router mode Range 0 4096 default is 1024 Typically one port per connection Grandstream Networks Inc HT503 User Manual Firmware 1 0 0 9 Page 20 of 37 Last Updated 9 2007
56. re sent based on the following events product model version on boot up INFO level NAT related info INFO level sent or received SIP message DEBUG level SIP message summary INFO level inbound and outbound calls INFO level registration status change INFO level negotiated codec INFO level Ethernet link up INFO level SLIC chip exception WARNING and ERROR levels e memory exception ERROR level The Syslog uses USER facility In addition to standard Syslog payload it contains the following components GS_LOG device MAC address error code error message Ex May 19 02 40 38 192 168 1 14 GS_LOG 00 0b 82 00 a1 be 000 Ethernet link is up This is a special feature that enables the user to create a text file backup of your existing configuration Grandstream Networks Inc HT503 User Manual Page 23 of 37 Firmware 1 0 0 9 Last Updated 9 2007 AR Innovative IP Voice amp Video TABLE 10 HT503 FXS PORT SETTINGS PAGES DEFINITIONS Account Active SIP Server Outbound Proxy SIP Transport NAT Traversal STUN SIP User ID 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 Contact Validate incoming message SIP T1 Timeout When set to yes the FXS port is activated This field contains th
57. rence list that will be included with the same preference order in SDP message This defines the encoding rate for G723 vocoder Default setting is 6 3kbps This sets the iLBC size in 20ms or 30ms This defines the payload type for iLBC Default value is 97 The valid range is between 96 and 127 Defines payload type for G726 16 Default value is 100 Range is from 96 to 127 Defines payload type for G726 24 Default value is 99 Range is from 96 to 127 Defines payload type for G726 40 Default value is 103 Range is from 96 to 127 Defines payload type for G729E Default value is 102 Range is from 96 to 127 Default is No VAD allows detecting the absence of audio and conserves bandwidth by preventing the transmission of silent packets over the network 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 T 38 Auto Detect FolP by default or fax Pass Through must use PCMU PCMA Grandstream Networks Inc HT503 User Manual Firmware 1 0 0 9 Page 30 of 37 Last Updated 9 2007 Fax Tone Detection Mode Jitter Buffer Type Jitter Buffer Length SRTP Mode Caller ID Scheme Caller ID RX Level dB Caller ID Transport Type Hook Flash Timing Gain Enable Current Disconnect Current Disconnect Threshold ms Enable PSTN Disconnect Tone Detection PSTN Disconnect Tone AC Termination
58. rsal Plug and Play When set to Yes the HT503 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 Default is No This allows the user to change set a specific MAC address on the WAN interface Note Set in Hex format Sets the LAN subnet mask Default value is 255 255 255 0 Base IP for the LAN port which functions as default gateway for its LAN Default value is 192 168 2 1 The length of time the IP address is assigned to the LAN clients Value is set in units of hours Default value is 120 hrs 5 days This function forwards all WAN IP traffic to a specific IP address if no matching port is used by HT503 or in the defined 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 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 PIN code to bridge from VoIP to PSTN Maximum 8 digits No Default PIN code to bridge from PSTN to VoIP Maximum 8 digits No Default
59. s Gateway IP address DNS Server IP address Preferred Vocoder WAN Port Web Access Firmware Server IP Address Configuration Server IP Address Upgrade Protocol Firmware Version Firmware Upgrade Direct IP Calling RESET Invalid Entry GC Innovative IP Voice amp Video Press for the next menu option Press to return to the main menu Enter 01 05 07 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 e PCMU PCMA G 723 G 729 G 726 iLBC 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 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
60. ss of the LAN port http 192 168 2 1 You will see the log in page of the device e 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 e 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 e Initial access to the configuration pages is always from the LAN port The instructions are listed above e 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 END USER CONFIGURATION 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 Password Web pages allowed End User Level 123 Only Status and Basic Settings Administrator Level admin Browse all pages Only an administrator can access the ADVANCED SETTING configuration page Please reference the GUI pages using the following link http Awww grandstream com user_manuals GUI GUI_ HT503 rar Once this HTTP request is entered and sent from a Web browser the HT503 will respond with the following login screen Grandstream Networks Inc HT503 User Manual Page 18 of 37 Firmware 1 0
61. standard Only use the universal power adapter provided with the HT503 package The manufacturer s warranty does not cover damages to the phone caused by unsupported power adaptors WARRANTY If you purchased your HT503 from a reseller please contact them 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 e This document is contains links to Grandstream GUI Interfaces Please remember to download these examples http www grandstream com user_manuals GUI GUI_HT503 rar for your reference e 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 user_manuals HT503 User Manual pdf e 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 Grandstream Networks Inc HT503 User Manual Page 4 of 37 Firm
62. ted from the remote side User can pre configure this tone on the ATA The user should know the frequency values and cadences of these tones Here is an example for the syntax for a busy tone in the U S A Syntax f1 freq vol f2 freq vol c on1 off1 on2 off2 on3 off3 Note freq 0 4000HZ vol 30 0dBm Default Busy Tone f1 480 24 f2 620 24 c 500 500 AC Termination is a configurable value of impedance of the line provided by different Grandstream Networks Inc HT503 User Manual Firmware 1 0 0 9 Page 31 of 37 Last Updated 9 2007 ndstrea Innovative IP Voice amp Video service providers in different countries 14 are selectable in this version of the F W Number of Rings Default is 4 This setting specifies number of phone rings on the phone connected to the FXS port before a PSTN incoming call is bridged to VoIP Note The number of rings feature serves as a PSTN answer delay and should be set to a larger value to allow enough time for the HT503 to decode the Caller ID signal set by the central office PSTN Ring Thru FXS If Yes the phone connected to the FXS port will ring a configured amount of times see above If not the phone connected to the FXS port will not ring 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
63. ter to SIP servers of ITSP Purposely left blank once saved for security Maximum length is 25 SIP service subscriber s name which will be used for Caller ID display Default is No If set to Yes the client will use DNS SRV to lookup for the SIP server If Yes is set a user phone parameter will be attached to the From header in SIP request This parameter controls whether the HT503 needs to send REGISTER messages to the proxy server The default setting is Yes Default is No If set to yes the device will first send registration request to remove all previous bindings Use only if proxy supports this remove bindings request This parameter allows users place outgoing calls even when not registered if allowed by ITSP but it s unable to receive incoming calls This parameter allows the user to specify the time frequency in minutes the HandyTone ATA 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 This parameter defines the local SIP port the HT503 will listen and transmit The default value for FXS port is 5060 This parameter defines the local RTP RTCP port pair used by the HandyTone ATA It is the base RTP port for channel 0 When configured the FXS port will use this port _value for RTP and the port_value 1 for its RTCP The default value for FXS port is 5004 Default is No If set to Yes the devic
64. the phone as the refresher or UAS to use the Callee or proxy server as the refresher As a Callee select UAC to use caller or proxy server as the refresher or UAS to use the phone as the refresher Session Timer can be refreshed using INVITE method or UPDATE method Select Yes to use INVITE method to refresh the session timer The HT503 supports 5 different Vocoder types including 1 G 711 Au law Displayed as PCMA PCMU 2 G 723 1 3 G 726 Supports bit rates 16 24 32 and 40 4 G 729A B E 5 iLBC Users can configure Vocoders in a preference list that will be included with the same preference order in SDP message Grandstream Networks Inc HT503 User Manual Firmware 1 0 0 9 Page 26 of 37 Last Updated 9 2007 G723 Rate iLBC Frame Size iLBC Payload Type G726 16 Payload Type G726 24 Payload Type G726 40 Payload Type G729E Payload Type VAD Symmetric RTP Fax Mode Fax Tone 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 Ring Tones Dag Innovative IP Voice amp Video This defines the encoding rate for G723 vocoder Default setting is 6 3kbps This sets the iLBC size in 20ms or 30ms This defines payload type for iLBC Default value is 97 The valid range is between 96 and 127 Defines payload type
65. traditional PBX you are using with HT503 uses this method for signaling call termination Default is No A configurable period of time in which the FXS port will drop off voltage on the line to indicate to the local party that the call is disconnected from the remote side 100 10000 ms Default 200 ms 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 Voice path volume adjustment e Rxis again level for signals transmitted by FXS e Txis again level for signals received by FXS Default OdB 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 or tone frequencies according to preference By default tones are set to North American frequencies Frequencies should be configured with known values to avoid high pitch sounds Grandstream Networks Inc HT503 User Manual Firmware 1 0 0 9 Page 2
66. utes to reduce the Firmware or Provisioning Server load at any given time Grandstream Networks Inc HT503 User Manual Page 36 of 37 Firmware 1 0 0 9 Last Updated 9 2007 ndstream Innovative IP Voice amp Video 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 VolP service provider FACTORY RESET IVR Command Reset default factory settings using the IVR Prompt Table 5 1 Dial for voice prompt 2 Enter 99 and wait for reset voice prompt 3 Enter the encoded MAC address Look below on how to encode MAC address 4 Wait 15 seconds and device will automatically reboot and restore factory settings Encoding 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 inthe 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 RESET Button Initiate the Factory Reset procedure by pressing the RESET button locat
67. waiting caller ID information which includes caller ID information in addition to the special stutter tone The analog phone must support this feature for it to work on the HT503 Both call waiting functions call waiting and CWCID are activated and deactivated from the configuration pages menu 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 1 A presses FLASH on the analog phone to hear the dial tone 2 Then A dials 87 then dials C s number and then presses 3 A can hang up NOTE Enable Call Feature has to be set to Yes in web configuration page Grandstream Networks Inc HT503 User Manual Page 11 of 37 Firmware 1 0 0 9 Last Updated 9 2007 ndstream Innovative IP Voice amp Video 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 The busy tone indicates the transfer has failed 3 Bus
68. ware 1 0 0 9 Last Updated 9 2007 ndstream Innovative IP Voice amp Video 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 VolP 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 FIGURE 1 CONNECTING THE HT503 HT503 Brees Back View Front View RJ 45 Ports Display LEDs 10 100 Mbps Green Power Supply 12V Reset 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 TABLE 1 DEFINITIONS OF THE HT503 CONNECTORS 12VDC 0 5A Power adapter connection LAN Port RJ 45 Connect the LAN port with an Ethernet cable to your PC WAN Port RJ 45 Connect the WAN port to the internal LAN network or router PHONE RJ 11 FXS port to be connected to analog phones fax machines LINE RJ 11 FXO port should b
69. 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 As a Callee select UAC to use caller or proxy server as the refresher or UAS to use the phone as the refresher Session Timer can be refreshed using INVITE method or UPDATE method Select Yes to use INVITE method to refresh the session timer The HT503 supports 5 different Vocoder types including 1 G 711 A p law Displayed as PCMA PCMU 2 G 723 1 3 G 726 Supports bit rates 16 24 32 and 40 4 G 729A B E 5 iLBC Users can configure Vocoders in a prefe
70. y tone keeps playing This means we have failed to receive the second NOTIFY from the transferee and the call has timed out Note this does not indicate the transfer has been successful nor does it indicate the transfer has failed When transferee is a client that does not support the second NOTIFY such as our own earlier firmware this situation occurs In bad network scenarios this could also happen although the transfer may have been completed successfully Attended Transfer This function is applicable on the FXS port for VolP calls only Assume that parties A and B are in conversation Party A wants to Attend Transfer Party B to C A presses FLASH on the analog phone to get a dial tone A then dial C s number followed by If C answers the call A and C are in conversation Then A can hang up to complete transfer If C does not answer the call A can press flash back to talk to B PP OAS NOTE When Attended Transfer fails and A hangs up the HT503 will ring user A back again to remind A that party B is still on the call Party A can pick up the phone to resume a conversation with party B 3 wAY CONFERENCING The HT503 supports both Star Code Style and Bellcore Style 3 way conferencing Star Code Style 3 way Conference This function is applicable on the FXS port for VoIP calls only Assume that parties A and B are in conversation Party A wants to bring C into a 3 way conference A presses FLASH on the analog phone or Hoo

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