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1. 23 Blige cc 24 SAVING THE CONFIGURATION 43 REBOOTING FROM REMOTE 43 CONFIGURATION THROUGH A CENTRAL SERVER cccceccccccecceecceececcecececeecesecsececceeaueeeneceeanees 43 SOFTWARE 44 FIRMWARE UPGRADE THROUGH ttt tnnt 44 CONFIGURATION FILE DOWNLOAD 45 FIRMWARE AND CONFIGURATION FILE PREFIX AND POSTFIX 45 MANAGING FIRMWARE AND CONFIGURATION FILE DOWNLOAD 45 RESTORE FACTORY DEFAULT 47 FIRMWARE VERSION 1 0 6 8 HT503 USER MANUAL Page 2 of 48 innovative IP Voice amp Video TABLE OF FIGURES HT503 UsER MANUAL Figure 1 CONNECTING THE HT503 sese ren 9 Figure 2 INTERCONNECTION DIAGRAM OF THE HT503 eee 10 Figure 3 UPLINK DOWNLINK BANDWIDTH LIMITATION eere 27 TABLE OF TABLES HT503 UsER MANUAL Table 1 DEFINITIONS OF THE HT508 CONNECTORS seen 9 Table 2 HT503 LED DEFINITIONS
2. aada an 9 Table 3 HT503 TECHNICAL SPECIFICATIONS sese 11 Table 4 HT503 HARDWARE SPECIFICATION sess enne nnne 12 Table 5 HT503 IVR MENU DEFINITIONS ssssssseseseeeenee enne 13 Table 6 HT503 CALL FEATURE DEFINITIONS essen nennen nennen 21 Fable 7 STATUS ie EE 24 Table 8 BASIC SETTINGS 25 Table 9 ADVANCED SETTINGS 28 Table 10 FXS PORT SETTINGS sse eene 31 Table 11 FXO PORT Settings niasin enanada iaaea asiaasi a aN a erada 37 TABLE OF GUI INTERFACES HT503 USER MANUAL http www grandstream com products ht _series ht503 documents ht503_gui zip SCREENSHOT OF CONFIGURATION LOGIN PAGE SCREENSHOT OF STATUS PAGE SCREENSHOT OF BASIC SETTINGS CONFIGURATION PAGE SCREENSHOT OF ADVANCED SETTINGS CONFIGURATION PAGE SCREENSHOT OF FXS ACCOUNT CONFIGURATION SCREENSHOT OF FXO ACCOUNT CONFIGURATION SCREENSHOT OF CALL PROGRESS TONES CONFIGURATION PAGE SCREENSHOT OF SAVED CONFIGURATION CHANGES SCREENSHOT OF REBOOT PAGE N O FIRMWARE VERSION 1 0 6 8 HT503 USER MANUAL Page 3 of 48 innovative IP Voice amp Video GNU GPL INFORMATION HT503 firmware contains third party software licensed under the GNU General Public License GPL Grandstream uses software under th
3. at least 2 digits number e atleast 1 digit number FIRMWARE VERSION 1 0 6 8 HT503 USER MANUAL Page 33 of 48 Subscribe for MWI Send Anonymous Anonymous Call Rejection Special Feature Session Expiration ndstream Innovative IP Voice amp Video e exclude e 8 5 any digit of 3 4 or 5 e 147 any digit 1 4 or 7 e 2 011 replace digit 2 with 011 when dialing e lt 1 gt add a leading 1 to all numbers dialed vice versa will remove a 1 from the number dialed e or Example 1 369 11 1617 Allow 311 611 911 and any 10 digit numbers of leading digits 1617 e Example 2 1900x lt 1617 gt Block any number of leading digits 1900 and add prefix 1617 for any dialed 7 digit numbers Example 3 1 2 9 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 1900 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 1900x 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 161
4. DNS Server IP address Preferred Vocoder MAC Address WAN Port Web Access Firmware Server IP Address Configuration Server IP Address Upgrade Protocol Firmware Version Firmware Upgrade Press for the next menu option Press ff to return to the main menu Enter 01 05 07 10 12 17 47 or 99 menu options Press 9 to toggle the selection If using Static IP configure the IP address information using menus 02 to 05 If using Dynamic IP Mode all IP address information comes from the DHCP server automatically after reboot The current WAN IP address is announced If using Static IP Mode enter 12 digit new IP address You need to reset the HT to take affect the new IP address Same as menu 02 Same as menu 02 Same as menu 02 Press 9 to move to the next selection in the list iLBC G 726 G 723 G 729 Announces the MAC address 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 FIRMWARE VERSION 1 0 6 8 HT503 U
5. 1 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 If hangs up the conference will be terminated for all three parties when configuration Transfer on Conference Hangup is set to No If the configuration is set to Yes A will transfer B to C so that B and C can continue the conversation 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 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 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 init
6. Force INVITE Preferred Vocoder Voice Frames per TX ndstream Innovative IP Voice amp Video 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 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 Default is No If set to Yes device will send an INVITE with audio vocoders upon completition of Fax to continue session in audio only For fax machines that do not send a D
7. No Key Entry Timeout Early Dial Dial Plan Prefix Use as Dial key Dial Plan ndstream Innovative IP Voice amp Video Used to replace SIP User Agent Header No Default Custom Ring Tone 1 to 3 with associate Caller ID when selected if Caller ID is configured then the device will ONLY uses this ring tone when the incoming call is from the Caller ID System Ring Tone is used for all other calls When selected but no Caller ID is configured the selected ring tone will be used for all incoming calls Distinctive ring tones can be configured not only for matching whole number but also for matching prefixes In this case symbol star will be used If server supports Alert Info header and standard ring tone set Bellcore or distinctive ring tone 1 10 is specified then the ring tone in the Alert Info header from server will be used For example If configured as 617 Ring Tone 1 will be used in case of call arrived from Massachusetts Any other incoming call will ring using cadence defined in parameter System Ring Cadence located under Advanced Settings Configuration page Default is No Default is No This is to disable the caller ID when a call waiting information arrives Default is No This is to disable the stutter Call Waiting Tone when a Call Waiting information arrives The CWCID information will still be displayed Default is No The reminder ring for the on hold call will not be played when this is set to Yes
8. dstream Innovative IP Voice amp Video Grandstream Networks Inc HT503 FXS FXO Port Analog Telephone Adaptor HT503 USER MANUAL dstream Innovative IP Voice amp Video HT503 User Manual Index GNU GPL 4 yl GE ECOU UG X 5 CHANGES FROM 1 0 5 10 USER _ 5 WELCOME 6 iejidumAeen lle 6 l uniinnandeee 6 CONNECT YOUR 8 EQUIPMENT PACKAGING 8 CONNECTING THE 503 8 PRODUCT OVERVIEW erreur nnno runner nra run 11 SOFTWARE FEATURES OVERVIEW eeseeeeee mI mnn Innen ern 11 HARDWARE SPECIFICATION 12 BASIC OPERATIONS 13 UNDERSTANDING HT503 VOICE PROMPT cccccccccceccccccecccccccececuececcececcecaececueeaueeenseeeaueecaecenaes 13 PLACING A PHONE
9. 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 fL 911x This shows that only calls that start with 911 are immediately forwarded to the PSTN line All other numbers will not be routed to the PSTN An normal would be L 617x x or x L 617x For example if Route Call to PSTN is configured as L 626x all outgoing calls starting with 626 will be initiated from the PSTN line 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 incomi
10. Block any number of leading digits 1900 and add prefix 1617 for any dialed 7 digit numbers Example 3 1 2 9 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 6 Default Outgoing x 92ooopp 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 1900x prevents dialing any number started with 1900 e 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 011 2 9 x allows international calls starting with 011 FIRMWARE VERSION 1 0 6 8 HT503 USER MANUAL Page 39 of 48 Subscribe for MWI Anonymous Call Rejection Special Feature Session Expiration Min SE Caller Request Timer Callee Request Timer Force Timer UAC Specify Refresher UAS Specify Refresher Force INVITE Invite Ring No Answer Timeout Enable 100rel Preferred Vocoder Voice frame per TX ndstream Innovative IP Voice amp Video 3469 11 allow dialing special and emergency numbers 311 411 611 and 911 N
11. 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 username Necessary if your ISP requires you to use PPPoE Point to Point Protocol over Ethernet connection PPPoE account password 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 FIRMWARE VERSION 1 0 6 8 HT503 USER MANUAL Page 25 of 48 Self Defined Time Zone Language Device Mode NAT Maximum Ports 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 ndstream Innovative IP Voice amp Video 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 i
12. 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 the key can be included as part of a number Dial plans work only for incoming calls from PSTN network In case unconditional call forward to VoIP is configured dial plan feature will not work In case of normal dialing to VoIP after dialing PSTN number If using the hop on hop off feature the dial plan rules affect only the last called number i e the number called after receiving dial tone from the ATA Dial Plan Rules 4 Accept Digits 1 2 3 4 5 6 7 8 9 0 A a B b C c D d 5 Grammar x any digit from 0 9 at least 2 digits number xx at least 2 digits number 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 Example 1 369 11 1617xxxxxxx Allow 311 611 911 and any 10 digit numbers of leading digits 1617 Example 2 1900 lt 1617 gt
13. URI SIP Registration Unregister on Reboot Outgoing Call w o Registration Register Expiration ndstream Innovative IP Voice amp Video Table 10 FXS PORT SETTINGS When set to yes the FXS port is activated This field contains the 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 This Field contains the URL or the IP address of a second SIP server this one will be used in case the device loses the connection with the first server 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 Thi
14. a valid PIN if it is invalid the HT503 will hang up 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 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 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 FIRMWARE VERSION 1 0 6 8 HT503 USER MANUAL Page 18 of 48 Innovative Voice amp Video e On the web configuration page if the Forward to VolP 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
15. 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 be connected to the PSTN line Table 2 HT503 LED DEFINITIONS POWER LED Indicates Power Remains ON when power is connected WAN LED Indicates LAN or WAN port activity LAN LED Indicates PC or LAN port activity PHONE LINE LED Indicates the status of the FXS and FXO ports on the back panel Busy ON Solid Green Available OFF Slow blinking FXS LEDs indicates voicemail for that port Note Slow blinking of POWER WAN and LAN LEDs together indicate firmware upgrade provisioning state FIRMWARE VERSION 1 0 6 8 HT503 USER MANUAL Page 9 of 48 ream Innovative IP Voice amp Video 9 D Internet ADSL Cable Modem Ethernet Tt Analog Phone Cordless Figure 2 INTERCONNECTION DIAGRAM OF THE HT503 FIRMWARE VERSION 1 0 6 8 HT503 USER MANUAL Page 10 of 48 dstream Inn
16. number of voice frames to be transmitted in a single packet When setting this value the user should be aware of the requested packet time used in SDP message as a result of configuring this parameter This parameter is associated with the first vocoder in the above vocoder Preference List or the actual used payload type negotiated between the 2 conversation parties at run time FIRMWARE VERSION 1 0 6 8 HT503 USER MANUAL Page 40 of 48 G723 Rate iLBC Frame Size iLBC Payload Type AAL2 G726 16 Payload Type AAL2 G726 24 Payload Type AAL2 G726 32 Payload Type AAL2 G726 40 Payload Type VAD Symmetric RTP Fax Mode Fax Tone Detection Mode Jitter Buffer Type Jitter Buffer Length SRTP Mode Caller ID Scheme FSK Caller ID minimum RX Level dB FSK Caller ID Seizure Bits FSK Caller ID mark bits Caller ID Transport Type Hook Flash Timing Gain ndstream Innovative IP Voice amp Video Default is 2 from 1 to 4 for G711 G726 G729 only For example if this field is set to be 2 and if the first vocoder chosen is G729 or G711 or G726 then the ptime value in the SDP message of an INVITE request will be 20ms 2 x10ms If the configured voice frames per TX exceeds the maximum allowed value the ATA will not accept it and will use and save the precedent configured allowed value for the corresponding first vocoder choice This defines the encoding rate for G723 vocoder Default setting is 6 3kbps This set
17. 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 0OdBm Default Busy Tone f1 480 24 f2 620 24 c 500 500 Note Maximum supported cadences is 3 You can select the AC termination by Country or by Impedance 15 Countries are selectable in this version of the F W Select the Impedance used by the PSTN service provider 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 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 If the PSTN Ring Thru Delay is set to Yes all incoming PSTN calls through FXO will ring the phone connected to the FXS port after this delay or after caller id is detected whichever comes first 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 o
18. the flash button toggles between two active calls The HT503 also provides CWCID call 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 FIRMWARE VERSION 1 0 6 8 HT503 USER MANUAL Page 15 of 48 ia Innovative IP Voice amp Video 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 and B are in conversation Party A wants to Blind Transfer Party B to C 3 Apresses FLASH on the analog phone to hear the dial tone 4 Then A dials 87 then dials C s number and then presses 5 Acan hang up NOTE Enable Call Feature has to be set to Yes in web configuration page 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 i
19. time Updates the Network Time Protocol Values range from 5 1440 minutes 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 are sent based on the following events e product model version on boot up INFO level e related info INFO level e sentor received SIP message DEBUG level e SIP message summary INFO level e inbound and outbound calls INFO level e registration status change INFO level e negotiated codec INFO level e Ethernet link up INFO level e SLIC chip exception WARNING and ERROR levels 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 If Syslog is enabled and Send SIP Log is set to YES then SIP messages will also be delivered via Syslog Default is set to NO This is a special feature that enables the user to create a text file backup of your existing configuration FIRMWARE VERSION 1 0 6 8 HT503 USER MANUAL Page 30 of 48 Account Active SIP Server Failover SIP Server Outbound Proxy SIP Transport NAT Traversal STUN SIP User ID Authenticate ID Authentication Password Name DNS mode Tel
20. will allow the phone to enable session timer only when the remote party support this feature To turn off Session Timer select 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 Default is 40 seconds the range is between 5 and 300 seconds The use of the PRACK Provisional Acknowledgement method enables reliability to be offered to SIP provisional responses 1xx series This is very important if PSTN inter networking is to be supported A user s request to use reliable provisional responses is invoked by the 100rel tag which is appended to the value of the required header of initial signalling messages The 503 supports up to 5 different Vocoder types including G 711 A U law G 726 Supports bit rates 16 24 32 and 40 G 723 1 G 729A B E and iLBC The user can configure Vocoders in a preference list that will be included with the same preference order in SDP message The first Vocoder is entered by choosing the appropriate option in Choice 1 The last Vocoder is entered by choosing the appropriate option in Choice 8 This field contains the
21. 14 PHONE OR EXTENSION NUMBERS 14 DIRECT IP CALES a Pa Ra E 14 G7 B INI OG B DRAMA 15 CALL 15 CALL TRANSFER irse sas nnana anaana dena 16 3 WAY CONFERENCING 16 PSTN PASS THROUGH 17 VOIP TO PSTN 17 PSTN TO VOIP 18 ROUTE CALLES a 19 FORWARD CALLS PSTN 19 FORWARD CALLS VOIP ccccccccccescccececccccccceueccccccecucecsuceeauceuueceuceeaueeuaucesaueeuaecensceeeueeuausenaueeaecens 19 ONE STAGE DIAL IN Gi clavate 19 FAX cls aan 20 CALL 21 CONFIGURATION 22 CONFIGURING HT503 THROUGH VOICE PROMPT 22 CONFIGURING HT503 WITH WEB BROWSER
22. 7 Defines payload type for AAL2 G726 16 Default value is 100 Range is from 96 to 127 Defines payload type for AAL2 G726 24 Default value is 99 Range is from 96 to 127 Defines payload type for AAL2 G726 24 Default value is 104 Range is from 96 to 127 Defines payload type for AAL2 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 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 e High initial 200ms min 40ms max 600ms Note not all vocoders can meet the high requirement e Medium initial 100ms min 20ms max 200ms Low initial 50 min 10ms max 100ms 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 ET
23. 7 area code will be added automatically 1 2 9 xx 2 9 xxxxxx allows dialing to any US Canada Number with 11 digits length 011 2 9 x allows international calls starting with 011 3469 11 allow dialing special and emergency numbers 311 411 611 and 911 Note some cases user wishes to dial strings such as 123 to activate voice mail or other application provided by service provider In this case should be predefined inside dial plan feature and the Dial Plan will be x 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 Asserted Identity 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 FIRMWARE VERSION 1 0 6 8 HT503 USER MANUAL Page 34 of 48 5 Caller Request Timer Callee Request Timer Force Timer UAC Specify Refresher UAS Specify Refresher Send Re INVITE After Fax Enable Silence Detection for Fax Disconnect Enable 100rel Use First Matching Vocoder in 2000K SDP
24. If set to YES the MWI information will not be transferred to the analog phone connected to the FXS port Sets the time in which an incoming call will stop ringing when not picked up Default value is 20 seconds In case this feature activated using codes 92 code the call will be forwarded after this preconfigured amount of time Default is 4 seconds Default is No Use only if proxy supports 484 response This parameter controls whether the phone will send an early INVITE each time a key is pressed when a user dials a number If set to Yes an INVITE is sent using the dial number collected thus far Otherwise no INVITE is sent until the Re Dial button is pressed or after about 5 seconds have elapsed The Yes option should be used ONLY if there is a SIP proxy configured and the proxy server supports 484 Incomplete Address response Otherwise the call will likely be rejected by the proxy with a 404 Not Found error Note This feature is NOT designed to work with and should NOT be enabled for direct IP to IP calling Sets the prefix added to each dialed number This allows users to configure the key as the Send or Dial key If set to Yes will send the number In this case this key is essentially equivalent to the Dial key If set to No the key can be included as part of a number Dial Plan Rules 1 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
25. 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 Both FXS port and FXO port are registered with this SIP Server FXS Port user has set Do Not Disturb 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 Table 8 BASIC SETTINGS This contains the password for end user to access the Web Configuration Menu User can put new password here This field is case sensitive with maximum of 25 characters 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 f 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 HT508 will attempt to establish a PPPoE session if PPPoE account is set f Static IP mode is selected the IP address Subnet Mask Default Router IP address
26. SER MANUAL Page 13 of 48 Innovative IP Voice amp Video 47 Direct IP Calling Enter the IP address to make a direct IP call after dial tone See Make a Direct IP Call 86 Voice Mail Number of voice mails 99 RESET Press 9 to reboot the device or Enter encoded MAC address to restore factory default setting See Restoring Factory Settings Invalid Entry Automatically returns to main menu NOTE e shifts down to the next menu option returns to the main menu 9 functions as the ENTER key in many cases to confirm an option e All entered digit sequences have known lengths 2 digits for menu option For IP address the key represent the dot Like 192 168 0 26 should be key in like 192 168 0 26 Once all of the digits are collected the input will be processed e Key entry cannot be deleted but the phone may prompt error once it is detected PLACING A PHONE CALL PHONE OR EXTENSION NUMBERS There are currently two methods to make an extension number call 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 seconds e dial a PSTN number such as 6266667890 you may need a prefix number followed by th
27. SI 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 Default 20dB If set to Yes polarity will be reversed upon call establishment and termination Default is No Set it to Yes of the 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 FIRMWARE VERSION 1 0 6 8 HT503 USER MANUAL Page 36 of 48 On Hook Timing Gain Disable Line Echo Canceller LEC Ring Tones Account Active SIP Server Failover SIP Server Prefer Primary SIP Server Outbound Proxy SIP Transport NAT Traversal STUN SIP User ID Authenticate ID Authenticate Password Name DNS mode ndstream Innovative IP Voice amp Video On hook timing is the minimum time for an on hook event to be validated Voice path volume adjustment e Rxis a gain level for signals transmitted by FXS Txis a gain level for signals received by FXS Default OdB for both parameters Loudest volume 6dB Lowest volume 6dB User can adjus
28. 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 followed 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 hook Will switch to a new channel for a new call Pressing pound sign will server as Re Dial key FIRMWARE VERSION 1 0 6 8 HT503 USER MANUAL Page 21 of 48 CC ia Innovativ
29. TN 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 HT500 GXV40XX UP Downlink Bandwidth Limitation by specified value in configuration or GUI RTP ip Nat Priority 2 Unlimited Bandwidth Bandwidth limited Figure 3 UPLINK DOWNLINK BANDWIDTH LIMITATION FIRMWARE VERSION 1 0 6 8 HT503 USER MANUAL Page 27 of 48 ndstream Innovative IP Voice amp Video Advanced User configuration includes not only the end user configuration but also advanced configurations such as SIP configuration Codec selection NAT Traversal Setting and other miscellaneous configuration Admin Password Layer 3 QoS Layer 2 QoS STUN Server Keep alive interval Use STUN to detect network activity Firmware Upgrade and Provisioning Via TFTP Via HTTP Via HTTPS Firmware Server Path Config Server Path XML Config File Password HTTP HTTPS User Name Table 9 ADVANCED SETTINGS Administrator password Only the administrator can configure th
30. a Ifthe target IP address is 192 168 0 160 the dialing convention is 47 or Voice Prompt with option 47 then 192 168 0 160 followed by pressing the key if it is configured as a send key or wait 4 seconds In this case the default destination port 5060 is used if no port is specified b Ifthe target IP address port is 192 168 1 20 5062 then the dialing convention would be 47 or Voice Prompt with option 47 then 192 168 0 160 5062 followed by pressing the 4 key if itis configured as a send key or wait for 4 seconds NOTE When completing direct IP call the Use Random Port should set to NO You can not make direct IP calls between FXS1 to FXS2 since they are using same IP CALL HOLD This function is applicable on the FXS port for VoIP calls only While in conversation pressing the flash button on the connected phone if the phone has that button places the remote end on hold Pressing the flash button again releases the previously held party and the conversation can resume If no flash button is available then on off hook quickly hook flash will do the same thing You may lose the call if hook flash is not quick enough 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
31. about 7 seconds Take out the pin All unit settings are restored to factory settings IVR Command Reset default factory settings using the IVR Prompt Table 5 Dial for voice prompt Enter 99 and wait for reset voice prompt Enter the encoded MAC address Look below on how to encode MAC address Wait 15 seconds and device will automatically reboot and restore factory settings Encode the MAC Address 1 Locate the MAC address of the device It is the 12 digit HEX number on the bottom of the unit 2 Keyin the MAC address Use the following mapping 0 9 0 9 22 press the 2 key twice A will show on the LCD 222 2222 33 press the 3 key twice D will show on the LCD 333 3333 For example if the MAC address is 000582006395 it should be keyed in as 0002228200333395 UIT EA OUT ae FIRMWARE VERSION 1 0 6 8 HT503 USER MANUAL Page 47 of 48 ia Innovative IP Voice amp Video NOTE 1 2 3 Factory Reset will be disabled if the Lock keypad update is set to Yes 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 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 Please be aware if the RESET button was pressed and release
32. ach particular field in the web configuration page A parameter consists of a Capital letter P and 2 to 3 Could be extended to 4 in the future digit numeric numbers i e P2 is associated with Admin Password in the ADVANCED SETTINGS page For a detailed parameter list please refer to the corresponding firmware release configuration template When Grandstream Device boots up or reboots it will issue request for configuration file named where is the LAN side MAC address of the device i e cfg000b820102ab The configuration file name should be in lower cases FIRMWARE AND CONFIGURATION FILE PREFIX AND POSTFIX Firmware Prefix and Postfix allows device to download the firmware name with the matching Prefix and Postfix This makes it possible to store ALL of the firmwares with different version in one single directory Similarly Config File Prefix and Postfix allows device to download the configuration file with the matching Prefix and Postfix Thus multiple configuration files for the same device can be stored in one directory In addition when the field Check New Firmware only when F W pre suffix changes is selected the device will only issue firmware upgrade request if there are changes in the firmware Prefix or Postfix MANAGING FIRMWARE AND CONFIGURATION FILE DOWNLOAD When Automatic Upgrade is set to Yes Service Provider can use P193 to have the devices perio
33. ains e One HT503 Main Case e One Universal Power Adaptor e One Ethernet Cable e One 503 Vertical Stand CONNECTING THE HT503 The HT503 is designed for easy configuration and easy installation Configure the HT503 following the directions in the Configuration section of this manual 1 Connect a standard touch tone analog telephone to the PHONE port 2 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 Connect a 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 The HT503 Analog Telephone Adaptor is an all in one VoIP integrated device designed to be a total solution for networks providing VoIP services The HT503 VoIP features and functions are available using a regular analog telephone FIRMWARE VERSION 1 0 6 8 HT503 USER MANUAL Page 8 of 48 dstream Innovative IP Voice amp Video HT503 HT503 Front View Back View Display LEDs RJ 45 Ports Power Green 10 100 Mbps Supply 12V Reset RJ11 RJ11 FXS Port FXO Port Figure 1 CONNECTING THE HT503 The HT503 has one FXS port and one FXO port The PHONE port next to the power supply is an FXS port The LINE port on the back right of the HT503
34. ait time Local SIP Port Local RTP Port Use Random Port Refer to Use Target Contact Remove OBP from Route Header Support SIP instance ID Validate incoming message Check SIP User ID for incoming INVITE SIP T1 Timeout SIP T2 Interval DTMF Payload Type Preferred DTMF method in listed order Disable DTMF Negotiation Proxy Require Use NAT IP Use SIP User Agent ndstream Innovative IP Voice amp Video SRV DNS SRV resource records indicates how to find services for various protocols NAPTR SRV Naming Authority Pointer according to RFC 2915 One mode can be chosen for the client to look up server The default value is A Record The default setting is Disabled If the phone has an assigned PSTN Number this field should be set to User Phone then User Phone parameter will be attached to the From header in the SIP request to indicate the E 164 number If server supports TEL URI format then this option needs to be selected 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
35. 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 FIRMWARE VERSION 1 0 6 8 HT503 USER MANUAL Page 31 of 48 Local SIP port Local RTP port Use Random Port Refer to Use Target Contact Transfer on conference hangup Enable Ring Transfer Disable Bellcore Style 3 Way Conference Remove OBP from Route Header Support SIP instance ID Validate incoming SIP message Check SIP User ID for incoming INVITE SIP T1 Timeout SIP T2 Interval DTMF Payload Type Preferred DTMF method in listed order Disable DTMF Negotiation Send Flash Event Enable Call Features Offhook Auto Dial Proxy Require Use NAT IP ndstream Innovative IP Voice amp Video 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 val
36. all 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 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 The FXO port functions as a bridge between the Internet and PSTN and enables calls to be passed from the PSTN network to VoIP The user can make VoIP calls remotely by dialing into the FXO line port on HT503 To Make a PSTN to VoIP Call 1 NOTE Make an incoming call to the PSTN line on FXO port The phone will ring for 4 times by default this setting is configurable on the FXO port configuration page 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 Enter a valid PIN if configured under the BASIC configuration page The caller will hear dial tone and be bridged to VoIP If an incorrect PIN is input the continuous tone prompts caller to enter a valid PIN The caller may try 3 times to enter
37. bs and auto switch to G 711 for Fax Pass through Fax Data pump V 17 V 19 V 27ter V 29 for T 38 fax relay HARDWARE SPECIFICATION The table below lists the hardware specification of HT503 Table 4 HT503 HARDWARE SPECIFICATION LAN interface 1xRJ45 10 100 Mbps Port WAN interface 1xRJ45 10 100 Mbps Port FXS telephone port 1x FXS RJ11 FXO telephone port PSTN Port 1x PSTN pass through and life line port LED Power WAN LAN PHONE and LINE Green Universal Switching Input 100 240 VAC 50 60 Hz Power Adaptor Output 12VDC 0 5A UL certified Dimension 25mm x 115mm x 75mm when laying flat 115mm x 25mm x 75mm standing up Weight Approximately 0 6lbs 0 3kg Temperature Operational 32 104 F or 5 45 C Storage 107 130 Humidity 10 90 non condensing Compliance FE FIRMWARE VERSION 1 0 6 8 HT503 USER MANUAL Page 12 of 48 ndstream Innovative IP Voice amp Video BASIC OPERATIONS UNDERSTANDING HT503 VOICE PROMPT HT503 has a built in voice prompt menu for simple device configuration The voice prompt menu is designed for the FXS port only To enter the voice prompt menu press from the analog phone connected to the FXS port Main Menu 01 02 03 04 05 07 10 12 13 14 15 16 17 Table 5 HT503 IVR MENU DEFINITIONS Enter a Menu Option DHCP Mode Static IP Mode IP Address IP address Subnet IP address Gateway IP address
38. ch the call will be rejected If this option is enabled the device will not be able to make direct IP calls T1 is an estimate of the round trip time between the client and server transactions If the network latency is high select larger value for reliable usage Maximum retransmission interval for non INVITE requests and INVITE responses Sends DTMF using RFC2833 The 503 supports up to different DTMF methods including in audio via RTP RFC2833 and via Sip Info User can configure DTMF method in a priority list Default is No If set to yes use above DTMF order without negotiation 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 Used to replace SIP User Agent Header No Default FIRMWARE VERSION 1 0 6 8 HT503 USER MANUAL Page 38 of 48 Header Ring Timeout Early Dial Dial Plan Prefix Use as Dial Key Dian Plan ndstream Innovative IP Voice amp Video 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
39. d in SDP message as a result of configuring this parameter This parameter is associated with the first vocoder in the above vocoder Preference List or the actual used payload type negotiated between the 2 conversation parties at run time Default is 2 from 1 to 4 for G711 G726 G729 only For example if this field is set to be 2 and if the first vocoder chosen is G729 or G711 or G726 then the ptime value in the SDP message of an INVITE request will be 20ms FIRMWARE VERSION 1 0 6 8 HT503 USER MANUAL Page 35 of 48 G723 Rate iLBC Frame Size iLBC Payload Type AAL2 G726 16 Payload Type AAL2 G726 24 Payload Type AAL2 G726 32 Payload Type AAL2 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 ndstream Innovative IP Voice amp Video 2 x10ms If the configured voice frames per TX exceeds the maximum allowed value the ATA will not accept it and will use and save the precedent configured allowed value for the corresponding first vocoder choice 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 12
40. d in less than 7 seconds the HT503 will only reboot it won t return to factory default settings FIRMWARE VERSION 1 0 6 8 HT503 USER MANUAL Page 48 of 48
41. d network settings codec settings and XML configuration settings FXS PORT To configure the FXS port e PORT To configure the port Table 7 STATUS PAGE MAC Address The device ID in HEX format This is very important ID for ISP troubleshooting Both LAN and WAN MAC addresses are located here The LAN MAC address is used for provisioning and is written on the label in the original box as well as on the label located on the back panel of the device WAN IP Address This field shows IP address of the HT503 Product Model This field contains the product model info such as HT503 Software Version Program This is the main software release This number is always used for firmware upgrade Current release is 1 0 6 8 Bootloader current version is 1 0 0 7 Core current version 1 0 5 9 Base current version is 1 0 6 8 System Uptime This shows system up time since last reboot Link Up This shows whether the PPPoE is up if connected to DSL modem FIRMWARE VERSION 1 0 6 8 HT503 USER MANUAL Page 24 of 48 Port Status 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 ndstream Innovative IP Voice amp Video This shows what kind of NAT the HT503 is connected to It is based on STUN protocol If the detected NAT is symmetric NAT STUN will not work and
42. dB Default 40dB Default is 70bits Range is from 0 to 800bits Default is 40bits Range is from 1 to 800bits 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 Identity 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 FIRMWARE VERSION 1 0 6 8 HT503 USER MANUAL Page 41 of 48 Enable Current Disconnect Current Disconnect Threshold ms Enable PSTN Disconnect Tone Detection PSTN Disconnect Tone AC Termination Model Country Based Impedance Based Number of Rings PSTN Ring Thru FXS PSTN Ring Thru Delay sec DTMF Digit Length ms DTMF Dial Pause ms First Digit Timeout sec ndstream Innovative IP Voice amp Video e RXis a gain level for signals transmitted by FXO To VoIP volume e TXis a gain level for signals received by FXO FXO To PSTN volume Default OdB for both parameters Loudest v
43. der 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 a FIRMWARE VERSION 1 0 6 8 HT503 USER MANUAL Page 20 of 48 02 03 16 17 30 31 47 50 51 67 82 69 70 71 72 73 78 79 87 90 91 92 93 Flash Hook ndstream Innovative IP Voice amp Video CALL FEATURES Table 6 HT503 CALL FEATURE DEFINITIONS Call Features Forcing a Codec per call 027110 PCMU 027111 PCMA 02723 G723 02729 G729 0272616 G726 r16 0272624 G724 r24 0272632 G726 r32 0272640 G726 r40 027201 iLBC Disable LEC pe call Dial 03 number No dial tone is played in the middle Enable SRTP Disable SRTP Block Caller ID for all subsequent calls Send Caller ID for all subsequent calls Direct IP Calling Dial 47 IP address No dial tone is played in the middle Detail see Direct IP Calling section on page 12 Disable Call Waiting for all subsequent calls 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
44. dically check with either Firmware Server or Config Server whenever they are defined This allows the device periodically check whether there is any new changes need to be taken similar to the AntiVirus Software to upgrade the Virus Definition files Screenshot is below FIRMWARE VERSION 1 0 6 8 HT503 USER MANUAL Page 45 of 48 ia innovative IP Voice amp Video Automatic Upgrade No Yes every 99 minutes 60 5256000 Yes daily at hour 923 Yes weekly on day 1 0 6 If automatic upgrade is enabled service provider can further customize the behavior and distribute server load by setting hour of the day and or day of the week for upgrade FIRMWARE VERSION 1 0 6 8 HT503 USER MANUAL Page 46 of 48 Cs ia 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 VoIP service provider FACTORY RESET There are two 2 methods for resetting your unit Reset Button Reset default factory settings following these four 4 steps 1 Unplug the Ethernet cable 2 Locate a needle sized hole on the back panel of the gateway unit next to the power connection 3 Insert a pin in this hole and press for
45. e Advanced Settings page Password field is purposely blanked for security reason after clicking update and saved The maximum password length is 25 characters This field defines the layer 3 QoS parameter which can be the value used for IP Precedence or Diff Serv or MPLS Default value is 48 Layer 2 QoS settings Default setting is blank VLAN supported equipment is required when configuring these settings IP address or Domain name of the STUN server This parameter specifies how often the HT503 sends a blank UDP packet to the SIP server in order to keep the NAT pin hole open Default is 20 seconds Use STUN keep alive to detect WAN side network problems If keep alive request does not yield any response for configured number of times the device will restart the TCP IP stack If the STUN server does not respond when the device boots up the feature is disabled Enables the HT503 to download firmware or configuration files through either TFTP or HTTP servers The default method is HTTP This is the IP address of the configured TFTP server If this is configured the HT503 retrieves the new configuration file or new code image from the specified TFTP server at boot time After 5 attempts the system will timeout and will start the boot process using the existing code image in the Flash memory If a TFTP server is configured and a new code image is retrieved the new downloaded image is saved into the Flash memory Note Firmware
46. e phone number Please check with your VoIP service provider for this information If your phone is assigned PSTN like number such as 6265556789 you will most likely follow the rule 1 the number 16266667890 Press or wait for 4 seconds DIRECT IP CALLS Direct IP calling allows two parties that is a FXS Port with an analog phone and another VoIP Device to talk to each other in an ad hoc fashion without a SIP proxy Elements necessary to completing a Direct IP Call e Both HT503 and other VoIP Device have public IP addresses or Both HT503 and other VoIP Device are on the same LAN using private IP addresses or e Both HT503 and other VoIP Device be connected through a router using public or private IP addresses with necessary port forwarding or DVZ FIRMWARE VERSION 1 0 6 8 HT503 USER MANUAL Page 14 of 48 Innovative IP Voice amp Video HT503 supports two ways to make Direct IP Calling Using IVR 1 Pick up the analog phone then access the voice menu prompt by dial 2 Dial 47 to access the direct IP call menu 3 Enter the IP address using format ex 192 168 0 160 after the dial tone Using Star Code 1 Pick up the analog phone then dial 47 2 Enter the target IP address using same format as above Note NO dial tone will be played between step 1 and 2 Destination ports can be specified by using encoding for followed by the port number Examples
47. e IP Voice amp Video CONFIGURATION GUIDE CONFIGURING HT503 THROUGH VOICE PROMPT DHCP Follow Table 4 with voice menu option 01 to enable HT503 to use DHCP STATIC Follow Table 4 with voice menu option 01 to enable HT503 to use STATIC IP mode then use option 02 03 04 to set up 503 5 IP Subnet Mask Gateway respectively FIRMWARE SERVER IP ADDRESS Select voice menu option 13 to configure the IP address of the firmware server CONFIGURATION SERVER IP ADDRESS Select voice menu option 14 to configure the IP address of the configuration server UPGRADE PROTOCOL Select voice menu option 15 to choose firmware and configuration upgrade protocol User can choose between TFTP HTTP and HTTPS FiRMWARE UPGRADE MODE Select voice menu option 17 to choose firmware upgrade mode There are three options 1 always check 2 check only when pre suffix changes and 3 never upgrade WAN Port WEB ACCESS Select voice menu option 12 to enable WAN Port Wed Access of the device configuration pages FIRMWARE VERSION 1 0 6 8 HT503 USER MANUAL Page 22 of 48 E itean Innovative IP Voice amp Video CONFIGURING HT503 WITH WEB BROWSER 503 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
48. e specific terms of the GPL Please see the GNU General Public License GPL for the exact terms and conditions of the license Grandstream GNU GPL related source code can be downloaded from Grandstream web site from http www grandstream com support fag gnu gpl FIRMWARE VERSION 1 0 6 8 HT503 USER MANUAL Page 4 of 48 innovative IP Voice amp Video CHANGE LOG This section documents significant changes from previous versions of HT503 user manuals Only major new features or major document updates are listed here Minor updates for corrections or editing are not documented here CHANGES FROM 1 0 5 10 USER MANUAL Add the option to change the Voice Frames per TX Voice Frames per TX Add a configuration parameter to override User Agent header Use SIP User Agent Header Added new Prompt Tone and Prompt Tone Access Code Prompt Tone Access Code Added Send SIP log in Syslog Send SIP Log Add NTP update interval option NTP Update Interval Added support for WebUI for Update and Apply Changes Changed the device design to accept parameters without requiring reboot FIRMWARE VERSION 1 0 6 8 HT503 USER MANUAL Page 5 of 48 Innovative 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
49. iate a call To MAKE A VoIP TO PSTN CALL 1 Dial the FXO SIP account phone number to establish the VoIP session The caller will hear the ring back tone once Then the caller hears either a special continuous tone or a dial tone The special continuous tone is played if the pin code is configured otherwise the caller will hear a dial tone Enter the PIN code if configured under the BASIC 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 three 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 FIRMWARE VERSION 1 0 6 8 HT503 USER MANUAL Page 17 of 48 Note ia Innovative IP Voice amp Video 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 there is no authentication required for callers on the use of PSTN line through HT503 When a PIN is configured for VOIP to PSTN c
50. isconnect when fax is done This option Enables Disables the detection of silence in order to know the fax has finished The silence period is non configurable and fixed to 7 seconds The use of the PRACK Provisional Acknowledgement method enables reliability to be offered to SIP provisional responses 1xx series This is very important if PSTN inter networking is to be supported A user s request to use reliable provisional responses is invoked by the 100rel tag which is appended to the value of the required header of initial signaling messages Default is No If set to Yes device will include only the first match vocoder in its 200 response otherwise it will include all match vocoders in same order received in INVITE 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 up to 5 different Vocoder types including G 711 A U law G 726 Supports bit rates 16 24 32 and 40 G 723 1 G 729A B E and iLBC The user can configure Vocoders in a preference list that will be included with the same preference order in SDP message The first Vocoder is entered by choosing the appropriate option in Choice 1 The last Vocoder is entered by choosing the appropriate option in Choice 8 This field contains the number of voice frames to be transmitted in a single packet When setting this value the user should be aware of the requested packet time use
51. least 512kbps limited for internal system signaling and NATed traffic Voice or RTP stream will never be limited See figure 3 The maximum downlink bandwidth permitted by the device This function is disabled by default The total bandwidth can be set as 128K 256K 512K 1M 2M 3M 4M 5M 10M or 15M The primary function of this setting is to limit the download bandwidth for the device internal system signaling and NATed traffic Example if 128 is configured there will be at least 128kbps limited for internal system signaling and NATed traffic Voice or RTP stream will never be limited See figure 3 When set to Yes the HT503 acts as an UPnP gateway for your UPnP enabled applications UPnP Universal 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 FIRMWARE VERSION 1 0 6 8 HT503 USER MANUAL Page 26 of 48 Address LAN DHCP Base IP LAN DHCP Start IP LAN DHCP End IP LAN Subnet Mask DHCP IP Lease Time DMZ IP Port Forwarding PSTN access code PIN for PSTN calls PIN for VoIP calls Unconditional Call Forward to PSTN Uncondi
52. n Browse all pages The password is case sensitive with maximum length of 25 characters The factory default password for End User and administrator is 123 and admin respectively Only an administrator can access the ADVANCED SETTING FXS PORT and FXO PORT configuration pages FIRMWARE VERSION 1 0 6 8 HT503 USER MANUAL Page 23 of 48 ndstream Innovative IP Voice amp Video NOTE If you cannot log into the configuration page by using the default password please check with the VoIP service provider It is most likely the VoIP service provider has provisioned the device and configured for you therefore the password has already been changed Only an administrator can access the ADVANCED SETTING FXS PORT and FXO PORT configuration pages Please reference the GUI pages using the following link http www grandstream com products ht_series ht503 documents ht503 gui zip DEFINITIONS This section will describe the options in the Web configuration user interface As mentioned a user can log in as an administrator or end user Functions available for the end user are STATUS Displays the network status account status software version and MAC address of the phone e BASIC SETTINGS Basic preferences such as date and time settings multi purpose keys and LCD settings can be set here Additional functions available to administrators are ADVANCED SETTINGS To set advance
53. nd 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 support firmware FIRMWARE VERSION 1 0 6 8 HT503 USER MANUAL Page 44 of 48 Innovative IP Voice amp Video Instructions for local firmware upgrade 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 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 Startthe 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 End users can also choose to download the free HTTP server from http httpd apache org or use Microsoft IIS web server CONFIGURATION FILE DOWNLOAD Grandstream SIP Device can be configured via Web Interface as well as via Configuration File through TFTP or HTTP HTTPS Config Server Path is the TFTP or HTTP HTTPS server path for configuration file It needs to be set to a valid URL either in FQDN or IP address format The Config Server Path be same or different from the Firmware Server Path A configuration parameter is associated with e
54. ndicates the transfer has failed 3 Busy 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 VoIP 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 Reo qe If C does not answer the call A can press flash back to talk to B 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 Bellcore Style 3 way conferencing FIRMWARE VERSION 1 0 6 8 HT503 USER MANUAL Page 16 of 48 itean Innovative IP Voice amp Video Assume that parties A and B are in conversation Party A using the HT503 wants to bring C into a 3 way conference
55. ng 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 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 dial directly to a local PSTN number if the one stage dialing feature is activated This feature is configured under the FXO FIRMWARE VERSION 1 0 6 8 HT503 USER MANUAL Page 19 of 48 Innovative IP Voice amp Video 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 provi
56. 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 address in 192 168 2 x segment by default 5 Open a web browser type in the default IP address of the LAN port http 192 168 2 1 You will see the log in page of the device FROM THE WAN PORT 1 Follow table 4 to find the WAN side IP address 2 Open a web browser type in the WAN side IP address for example http HT503 WAN IP Address Note WAN side HTTP access is disabled by default for security reason You can enable HTTP access on the configuration page by setting WAN side HTTP access to be YES Initial access to the configuration pages is always from the LAN port The instructions are listed above IVR announces 12 digits IP address you need to strip out the leading in the IP address For ex IP address 192 168 001 014 you need to type in http 192 168 1 14 in the web browser Once the HTTP request is entered and sent from a web browser the user will see a log in screen There are two default passwords for the login page User Level Password Web pages allowed End User Level 123 Only Status and Basic Settings Administrator Level admi
57. olume 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 disconnected from the remote side User can pre configure this tone on
58. ote In some cases user wishes to dial strings such as 123 to activate voice mail or other application provided by service provider In this case should be predefined inside dial plan feature and the Dial Plan will be x Default is No When set to Yes a SUBSCRIBE for Message Waiting Indication will be sent periodically Default is No If set to Yes incoming calls with anonymous Caller ID will be rejected with a 486 busy message Default is Standard Choose the selection to meet some special requirements from Softswitch vendors Grandstream implemented SIP Session Timer The session timer extension enables SIP sessions to be periodically refreshed via a SIP request UPDATE or re INVITE Once the session interval expires if there is no refresh via a UPDATE or re INVITE message the session will be terminated Session Expiration is the time in seconds at which the session is considered timed out if no successful session refresh transaction occurs beforehand The default value is 180 seconds The minimum session expiration in seconds The default value is 90 seconds If selecting Yes the phone will use session timer when it makes outbound calls if remote party supports session timer If selecting Yes the phone will use session timer when it receives inbound calls with session timer request If selecting Yes the phone will use session timer even if the remote party does not support this feature Selecting No
59. ovative 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 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 SOFTWARE FEATURES OVERVIEW The HT503 features 2 SIP account profiles and supports advanced telephony features including caller ID call waiting call transfer 3 way conferencing with either IP or PSTN calls and multi language voice prompts From a technical standpoint the HT503 offers a power outage survivable life line and 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 3 HT503 TECHNICAL SPECIFICATIONS Interfaces 1 FXS telephone port RJ11 1 FXO PSTN line port RJ11 with lifeline support Two 2 10M 100 Mbps ports RJ45 with integrated Nat router Protocol Support TCP UDP IP RTP RTCP HTTP HTTPS ARP RARP ICMP DNS DHCP NTP TFTP PPPoE STUN amp TELNET
60. protocols CA PE M S LED Indicators Power WAN LAN PHONE and LINE Device Management Web interface or via secure AES encrypted central configuration file for mass Support device configuration via built in through TFTP HTTP or HTTPS Support Layer 2 802 1Q VLAN 802 1 and Layer QoS ToS DiffServ MPLS friendly remote software upgrade via including behind firewall NAT A d E ETT TT ETT ETT TT eT ene E ere open ert ener ent ener en peer Syslog support DHCP Server Client Yes inscr ates ba Audio Features Dynamic negotiation of codec and voice payload length Silence Suppression VAD voice activity detection CNG comfort noise generation ANG automatic gain control FIRMWARE VERSION 1 0 6 8 HT503 USER MANUAL Page 11 of 48 dstream Innovative IP Voice amp Video auus d Ic uc aaa ee p CD ppc D c 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 n MN Network and Manual dynamic host configuration protocol DHCP network setup NAT Provisioning Support traversal via Fax over T 38 compliant Group 3 Fax Relay up to 14 4kp
61. rd ACS URL ACS Username ACS Password Periodic Inform Enable Periodic Inform Interval Connection Request Username Connection Request Password System Ring Cadence Call Progress Tones ndstream Innovative IP Voice amp Video The password for the HTTP HTTPS 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 If set to Yes configuration and upgrade server information can be obtained using DHCP option 66 from DHCP server located in customer s environment Choose Yes to enable automatic upgrade and provisioning When set to No HT503 will only do upgrade once at boot up When Check every day or Check every week is checked user can specify Hour of the day 0 23 or Day of the week 0 6 Default time is Monday 1AM There are three options to choose from Always check for New Firmware at Boot up Check New Firmware only when F W pre suffix changes and Always Skip the Firmware Check This protects the configuration from an unauthorized change If set to Yes the configuration file is authenticated before acceptance Key for firmware encryption 32 digits in hexadecimal format End users should keep it blank The user specified SSL ce
62. rder 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 Dial pause is the time between 2 digits for the same scenario as explained above 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 FIRMWARE VERSION 1 0 6 8 HT503 USER MANUAL Page 42 of 48 dstream Innovative IP Voice amp Video 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 SAVING THE CONFIGURATION CHANGES After user makes a change to the configuration press the Update button in the Configuration Menu The web browser will then display a message window to confirm saved changes press Apply button to confirm Grandstream recommends reboot or power cycle the IP phone after saving changes REBOOTING FROM REMOTE Press the Reboot button at the bot
63. recommend to maintain their own TFTP HTTP HTTPS server for upgrade and provisioning procedures Once a Firmware Server Path is set user needs to update the settings and reboot the device If the configured firmware server is found and a new code image is available the HT503 will attempt to retrieve the new image files by downloading them into the HT503 s SRAM During this stage the HT503 s 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 TFTP HTTP HTTPS fails for any reason e g TFTP HTTP HTTPS server is not responding there are no code image files available for upgrade or checksum test fails etc the HT503 will stop the TFTP HTTP HTTPS process and simply boot using the existing code image in the flash Firmware upgrade may take as long as 15 to 30 minutes over Internet or just 5 minutes if it is performed on a LAN t is recommended to conduct firmware upgrade in a controlled LAN environment if possible For users who do not have a local firmware upgrade server Grandstream provides a NAT friendly HTTP server on the public Internet for firmware upgrade Grandstream s latest firmware is available http www grandstream com support firmware Oversea users are strongly recommended to download the binary files and upgrade firmware locally in a controlled LAN environment Alternatively user can download a free TFTP or HTTP server a
64. repair or refund If you purchased the product directly from Grandstream contact your Grandstream Sales and Service Representative for an 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 avoid your manufacturer warranty FIRMWARE VERSION 1 0 6 8 HT503 USER MANUAL Page 6 of 48 dstream Innovative IP Voice amp Video e This document contains links to Grandstream GUI Interfaces Please remember to download these examples from http Awww grandstream com products ht_series ht503 documents ht503 gui zip 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 products ht series htb03 documents ht503 usermanual english 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 FIRMWARE VERSION 1 0 6 8 HT503 USER MANUAL Page 7 of 48 Cs ia innovative IP Voice amp Video CONNECT YOUR HT503 EQUIPMENT PACKAGING The HT503 ATA package cont
65. round trip time between the client and server transactions If the network latency is high select larger value for more reliable usage Maximum retransmission interval for non INVITE requests and INVITE responses This parameter sets the payload type for using RFC2833 The 503 supports up to different DTMF methods including in audio via RFC2833 and via Sip Info The user can configure DTMF method in a priority list Default is No If set to yes use above DTMF order without negotiation Default is No If set to yes flash will be sent as DTMF event Default is Yes If Yes call features using star codes will be supported locally This parameter allows users to configure a User ID or extension number 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 FIRMWARE VERSION 1 0 6 8 HT503 USER MANUAL Page 32 of 48 Use SIP User Agent Header Distinctive Ring Tone Disable Call Waiting Disable Call Waiting Caller ID Disable Call Waiting Tone Disable Reminder Ring for On Hold Call Disable Visual MWI Ring Timeout
66. router performance than its predecessor the HT488 It is the second in the HandyTone 50x series The HT503 functions as a true 3 in 1 gateway for PSTN network analog telephone FXS interface and IP network It enables remote call origination and termination from to PSTN and supports the feature of hop on hop off calling This manual will help you learn how to operate and manage your HT503 Analog Telephone Adaptor IAD and make the best use of its many upgraded features including simple and quick installation 3 way conferencing and remote call origination and hop on hop off calling using the programmable PSTN FXO port This HT503 is very easy to manage and configure and is specifically designed to be an easy to use and affordable VoIP solution for both the residential user and the remote user This document is subject to changes without notice The latest electronic version of this user manual can be downloaded from the following location http www grandstream com products ht_series ht503 documents ht503_ usermanual english pdf SAFETY COMPLIANCS The HT503 adaptor complies with FCC CE and various safety standards The HT503 power adaptor is compliant with UL standard Only use the universal power adapter provided with the HT503 package The manufacturers 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
67. rtificate 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 User specify the Auto Configuration Servers URL TR 069 protocol User specify the ACS Username User specify the ACS password Default is No If set to YES device will send inform packets to the ACS Frequency that the inform packets will be sent out to the ACS Set a user name for the ACS to connect to this device Set a password for the ACS to connect to this device Configuration option for FXS port ring cadence for all incoming calls Syntax c on1 off1 on2 off2 on3 off3 Note Maximum supported cadences is 3 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 112350 9 13 12 440 0 13 c 0 0 FIRMWARE VERSION 1 0 6 8 HT503 USER MANUAL Page 29 of 48 Access Code Lock Keypad Update Disable Voice Prompt Disable Direct IP Calling Life Line Mode NTP server NTP Update Inter
68. s field contains the user part of the SIP address for this phone e g if the SIP address is sip my user id omy 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 register 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 One from the 3 modes available for DNS Mode configuration A Record for resolving IP Address of target according to domain name SRV DNS SRV resource records indicates how to find services for various protocols NAPTR SRV Naming Authority Pointer according to RFC 2915 One mode can be chosen for the client to look up server The default value is A Record The default setting is Disabled If the phone has an assigned PSTN Number this field should be set to User Phone then a User Phone parameter will be attached to the From header in the SIP request to indicate the E 164 number If server supports TEL URI format then this option needs to be selected 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
69. s 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 for AAL2 G726 16 Default value is 100 Range is from 96 to 127 Defines payload type for AAL2 G726 24 Default value is 99 Range is from 96 to 127 Defines payload type for AAL2 G726 24 Default value is 104 Range is from 96 to 127 Defines payload type for AAL2 G726 40 Default value is 103 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 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 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 96 O
70. t is positive if the local time zone is west of the Prime Meridian and negative if itis 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 2 number indicates the nth iteration of the weekday 1 Sunday 3 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 NAT TCP idle timeout in seconds Connection will be closed after preconfigured timeout if not refreshed Range 0 3600 NAT TCP idle timeout in seconds Connection will be closed after preconfigured timeout if not refreshed Range 0 3600 default is 300 The maximum uplink bandwidth permitted by the device This function is disabled by default The total bandwidth can be set as 128K 256K 512K 1M 2M 3M 4M 5M 10M or 15M The primary function of this setting is to limit the uplink bandwidth for the device internal system signaling and NATed traffic Example if 512k is configured there will be at
71. t 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 Default is No If set to Yes LEC will be disabled per call base Recommended for FAX Data calls 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 Table 11 FXO PORT Settings When set to Yes the FXO port is activated SIP Server s IP address or Domain name provided by VoIP Service Provider This Field contains the URL or the IP address of a second SIP server this one will be used in case the device loses the connection with the first server Default is no If set to yes it will register to Primary Server if registration with Failover server expires 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
72. ted to customer s TFTP or HTTP server for further provisioning Grandstream also provide configuration tool to facilitate the task of generating device configuration files The tool and configuration template are available for download from http www grandstream com support tools FIRMWARE VERSION 1 0 6 8 HT503 USER MANUAL Page 43 of 48 Cs ia Innovative IP Voice amp Video SOFTWARE UPGRADE Software upgrade can be done via TFTP HTTP or HTTPS The corresponding configuration settings are in the ADVANCED SETTINGS configuration page FIRMWARE UPGRADE THROUGH TFTP HTTP HTTPS To upgrade via TFTP HTTP or HTTPS the Firmware Upgrade and Provisioning upgrade via field needs to be set to TFTP HTTP or HTTPS respectively Firmware Server Path needs to be set to a valid URL of a TFTP or HTTP server server name can be in either FQDN or IP address format Here are examples of some valid URL e g firmware mycompany com 6688 Grandstream 1 0 6 8 e g firmware grandstream com NOTES Firmware upgrade server in IP address format can be configured via IVR Please refer to the CONFIGURATION GUIDE section for instructions If the server is in FQDN format it must be set via the web configuration interface Grandstream recommends end user use the Grandstream HTTP server Its address can be found at http www grandstream com support firmware Currently the HTTP firmware server address is firmware grandstream com For large companies we
73. 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 503 will use its mapped public IP address and port in all of its SIP and SDP messages If 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 One from the 3 modes available for DNS Mode configuration A Record for resolving IP Address of target according to domain name FIRMWARE VERSION 1 0 6 8 HT503 USER MANUAL Page 37 of 48 Tel URI SIP Registration Unregister on Reboot Outgoing Call Without Registration Register Expiration SIP registration failure retry w
74. tional Call Forward to VoIP ndstream Innovative IP Voice amp Video Note Set in Hex format Base IP for the LAN port which functions as default gateway for its LAN Default value is 192 168 2 1 Note When the device detects WAN IP is conflicting with LAN IP the LAN base IP address will be changed based on the network mask the effective subnet will be increased by 1 For example 192 168 2 1 will be changed to 192 168 3 1 if net mask is 255 255 255 0 Then the device will reboot Default is 100 Default is 199 Sets the LAN subnet mask Default value is 255 255 255 0 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 HT508 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 OO 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 Calls are unconditionally forwarded to the specified PS
75. tom of the configuration menu to reboot the phone remotely The web browser will then display a message window to confirm that reboot is underway Wait 30 seconds to log in again CONFIGURATION THROUGH A CENTRAL SERVER Grandstream HT503 can be automatically configured from a central provisioning system When HT503 boot up it will send TFTP or HTTP HTTPS request to download configuration file cfg000b82 ooxxx or cfg00082 ooxxx xml where 000682 is the LAN MAC address of the HT503 It will first request cfg000b82xxxxxx then cfgO00b82xxxxxx xml A service provider or an enterprise with large deployment of Grandstream devices can easily manage the configuration and service provisioning of individual devices remotely from a central server Grandstream has a central provisioning system called GAPS Grandstream Automated Provisioning System GAPS supports automatic configuration of Grandstream devices GAPS uses enhanced NAT friendly TFTP or HTTP thus no NAT issues and other communication protocols to communicate with each individual Grandstream device Grandstream provides GAPS service to VoIP service providers Use GAPS for either simple redirection or with certain special provisioning settings At boot up Grandstream devices by default point to Grandstream provisioning server GAPS based on the unique MAC address of each device GAPS provision the devices with redirection settings so that they will be redirec
76. ue for FXS port is 5004 Default is No If set to Yes the device 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 In which case if conference originator hangs up the conference will be terminated When option YES is chosen originator will transfer other parties to each other so that B and C can choose either to continue the conversation or hang up Default is No this will create a Semi Attendant Transfer When set to Yes device can transfer the call upon receiving ring back tone Default is No you can make a Conference by pressing Flash key If set to Yes you need to dial 23 second callee number Default is No If set to Yes the Outbound Proxy will be removed from the route header Default is Yes If set to Yes the contact header in REGISTER request will contain SIP Instance ID as defined in IETF SIP Outbound draft Default is No If set to yes all incoming SIP messages will be strictly validated according to RFC rules If message will not pass validation process call will be rejected Default is No Check the incoming SIP User ID in Request URI If they don t match the call will be rejected If this option is enabled the device will not be able to make direct IP calls T1 is an estimate of the
77. 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 6 8 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 The URL of the HTTP server used for firmware upgrade and configuration via a secure HTTP connection For example https provisioning mycompany com Note the HTTPS default port is 443 IP address or domain name of firmware server IP address or domain name of configuration server The password used for encrypting the XML configuration file using OpenSSL This is required for the phone to decrypt the encrypted XML configuration file The user name for the HTTP HTTPS server FIRMWARE VERSION 1 0 6 8 HT503 USER MANUAL Page 28 of 48 HTTP HTTPS Password Firmware File Prefix Firmware File Postfix Config File Prefix Config File Postfix Allow DHCP option 66 to override server Automatic Upgrade Authenticate Conf File Firmware Key SSL Certificate SSL Private Key SSL Private Key Passwo
78. val Syslog Server Syslog Level Send SIP Log Download Device Configuration ndstream Innovative IP Voice amp Video Syntax f1 freq vol f2 freq vol c on1 off1 on2 off2 on3 off3 Note freq 0 4000Hz vol 30 0dBm Note Maximum supported cadences is 3 Key pattern to get Prompt Tone Maximum 20 digits No Default 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
79. with the specified registrar The default interval is 60 minutes or 1 hour The maximum interval is 65535 minutes about 45 days This parameters allows the user to specify the time frame in seconds the HT508 will wait before sending another SIP registration INVITE in case the first INVITE fails 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 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 the Outbound Proxy will be removed from the route header Default is Yes If set to Yes the contact header in REGISTER request will contain SIP Instance ID as defined in IETF SIP Outbound draft Default is No If set to yes all incoming SIP messages will be strictly validated according to RFC rules If message will not pass validation process call will be rejected Default is No Check the incoming SIP User ID in Request URI If they don t mat

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