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1. Users can choose whether apply password protection for PSTN to VoIP calls not PIN Pin to VoIP calls consists of up to 8 numeric digits can be configured through BASIC SETTINGS of the web configuration page By default there is no password protection 1 there is no authentication required for callers on the use of VoIP SIP account on FXO port Upon hearing the special continuous tone for PIN code input if the caller don t enter any digit HT488 will time out and hang up the call 10 seconds During any stage of digits input a 4 seconds timeout is applied to serve as an end of PIN or destination number input Users may also use the 4 key to indicate the end of an input On the web configuration page if the Forward to VoIP is configured the second stage dialing is eliminated 1 after bridging to VoIP the configured VoIP number will be called automatically 5 2 11 Route Calls to PSTN This function is applicable on FXO port that can access the PSTN network By default HT488 1s in VoIP mode upon off hook If Route call to PSTN 1 configured certain calls will be initiated from 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 prefix or a telephone number in the Route call to PSTN on BASIC SETTINGS web configuration page If the dialed digits match one of the specified prefix outbound calls wi
2. Automatic Gain Control Support standard encryption and authentication DIGEST using MD5 MD5 sess Support for Layer 2 802 10 VLAN 802 1p and Layer 3 QoS ToS DiffServ Support automated NAT traversal without manual manipulation of firewall NAT Support device configuration via built in IVR Web browser or central configuration file through TFTP or HTTP server Support firmware upgrade via or Support PSTN pass through on 2 0 Ultra compact wallet size and lightweight design great companion for travelers Compact lightweight Universal Power adapter HandyTone 488 User Manual Grandstream Networks Inc 4 2 Hardware Specification The table below lists the hardware specification of HandyTone 488 Model HandyTone 488 LAN interface 1xRJ45 10Base T WAN interface IxRJA45 10Base T FXS telephone port IxEXS FXO port Button 1 LED Green and red color Universal Switching Input 100 240 50 60 Hz Power Adaptor Output 5 VDC 1200mA UL certified Dimension 70mm W 130mm D 27mm H Weight 0 6lbs 0 3kg Temperature 40 130 F 545 Humidity 10 90 non condensing Compliance FE HandyTone 488 User Manual Grandstream Networks Inc 5 Basic Operations 5 1 Get Familiar with Voice Prompt HandyTone 488 has stored a voice prompt menu Interactive Voice Response or IVR for quick browsing and simple configuration To enter this voice prompt menu simply press the butt
3. H 323 etc 44
4. WAN side HTTP access to be YES The very first time to access the configuration page is always from LAN port The instructions are listed above e The IVR announces 12 digits IP address you need to strip out the leading O in address For ex address 192 168 001 014 you need to type http 192 168 1 14 in the web browser 6 2 2 End User Configuration Log in page Password Login 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 administrator can get access to ADVANCED SETTING configuration page Note Ifyou can not log into the configuration page by using default password please check with the voip service provider Most likely the service provider has provisioned the device and changed the login password Status Page 19 HandyTone 488 User Manual Grandstream Networks Inc MAC Address WAN IP Address Product Model Software Version System Up Time Registered PPPoE Link Up NAT MAC Address WAN IP Address Product Model Software Version System Uptime Registered PPPoE Link Up NAT 00 0B 82 03 DA 36 10 10 11 225 HT488 Program 1 0 3 18 Bootloader 1 0 8 9 HTML 1 0 3 18 VOC 1 0 0 10 0 day s 1 hour s 47 minute s Yes disabled detected NAT type 16 full cone The unique device ID in HEX format This
5. 01 to enable HandyTone 488 to use STATIC IP mode then use option 02 03 04 05 to set up HandyTone 488 s IP Subnet Mask Gateway and DNS server respectively 6 1 3 Server Address Follow section 5 1 with voice menu option 06 to configure the IP address of the TFTP server 6 2 Configuring HandyTone 488 with Web Browser HandyTone series 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 HandyTone 488 through a Web browser such as Microsoft s IE and AOL s Netscape 6 2 1 Access the Web Configuration Menu The HandyTone 488 HTML configuration page can be accessed via LAN or WAN port e From the LAN port Directly connect a computer to the LAN port Open a command window on the computer Type in ipconfig release the IP address etc becomes 0 Type in ipconfig renew the computer gets an IP address 192 168 2 x segment by default Open web browser type in the default gateway IP address You will see the log in page of the device http 192 168 2 1 e From the WAN port Follow section 5 1 to find out the WAN side IP address Open a web browser type in the WAN side IP address 18 HandyTone 488 User Manual Grandstream Networks Inc http HandyTone WAN IP Address Note e WAN side HTTP access is by default disabled for security reason You can enable it on the configuration page by setting
6. 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 if user needs to change these settings This parameter allows user to configure the key to be used as the Send or Dial key Once set to Yes pressing this key will immediately trigger the sending of dialed string collected so far If set to No the key will then be included as part of the dialed string to be sent out Defines whether the NAT traversal mechanism is activated It should be set to YES the device is behind NAT router If outbound proxy is not configured 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 every Keep alive interval send a dummy UDP packet to the SIP server to pinhole the NAT Default 1s 20 seconds The interval of sending dummy UDP packet to keep NAT pin hole open NAT IP address used in SIP SDP message Default 1s blank Default method is HTTP Firmware upgrade may take up to 10 minutes depending on network environment Do not interrupt the firmware upgrading process 25 HandyTone 488 User Manual Grandstream Networks Inc Firmware Server IP address or domain name of firmware server Path Config Server Path address or domain name of configuratio
7. HTTP thus no NAT issues and other communication protocols to communicate with each individual HandyTone ATA for firmware upgrade remote reboot etc Grandstream provide GAPS Grandstream Automated Provisioning System service to VoIP service providers It could be either simple redirection or with certain special provisioning settings Initially upon booting up Grandstream devices by default point to Grandstream provisioning server GAPS based on the unique MAC address of each device GAPS provision the devices with redirection settings so that they will be redirected to customer s HTTP server for further provisioning Grandstream also provide GAPSLite software package which contains our NAT friendly server and a configuration tool to facilitate the task of generating device configuration files The GAPSLite configuration tool 1s now free to end users The tool and configuration template are available for download from http www grandstream com DOWNLOAD Configuration_Tool For details on how GAPS works please contact Grandstream and refer to the documentation of GAPS product provided 35 HandyTone 488 User Manual Grandstream Networks Inc 7 Software Upgrade Software upgrade be done via either or The corresponding configuration settings are in the ADVANCED SETTINGS configuration page 7 1 Firmware Upgrade through To upgrade via TFTP or HTTP the Firmware Upgr
8. Telephony call are conversion of the analog voice signal to digital 41 HandyTone 488 User Manual Grandstream Networks Inc IVR MTU NAT NTP format and compression translation of the signal into Internet protocol IP packets for transmission over the Internet or other packet switched networks the process is reversed at the receiving end The terms IP Telephony and Internet Telephony are often used to mean the same however they are not 100 per cent interchangeable since Internet 15 only a subcase of packet switched networks For users who have free or fixed price Internet access IP Telephony software essentially provides free telephone calls anywhere in the world However the challenge of IP Telephony is maintaining the quality of service expected by subscribers Session border controllers resolve this issue by providing quality assurance comparable to legacy telephone systems IVR is a software application that accepts a combination of voice telephone input and touch tone keypad selection and provides appropriate responses in the form of voice fax callback e mail and perhaps other media A Maximum Transmission Unit MTU 1s the largest size packet or frame specified in octets eight bit bytes that can be sent in a packet or frame based network such as the Internet The maximum for Ethernet is 1500 byte Network Address Translation Network Time Protocol a protocol to exchange and synchronize time over networks The
9. anonymous essentially blocking the Caller ID from displaying Lock keypad If this parameter is set to Yes the configuration update via keypad 15 update disabled Special Features Default is Standard Choose the selection to meet some special requirements from Soft Switch vendors like Nortel Broadsoft etc FXS Impedance Selects the impedance of the analog telephone connected to the Phone port Caller ID Scheme Select the Caller ID Scheme to suit the standard of different area Bellcore North America CID Canada DTMF Brazil DTMF Denmark ETSI DTMF Finland Sweden ETSI FSK France Germany Norway Taiwan UK CCA Onhook Voltage Select the onhook voltage to suit the analog phone Polarity Reversal Select Polarity Reversal to adapt some call charge billing system Default 15 port page fI sip mycompany com IP address e g proxy myprovider com or IP address if any SIP User ID the user part of an SIP address 31 Outbound Proxy HandyTone 488 User Manual Authenticate ID Authenticate Password Name Use DNS SRV User ID is phone number SIP Registration Unregister On Reboot Register Expiration local SIP port local RTP port Use random port DTMF Payload Type Send DTMF Send Flash Event Proxy Require Preferred Vocoder in listed order Voice Frames per TX G723 rate iLBC frame
10. loop The DECT Common Interface radio standard is a multicarrier time division multiple access time division duplex MC TDMA TDD radio transmission technique using ten radio frequency channels from 1880 to 1930 MHz each 39 HandyTone 488 User Manual Grandstream Networks Inc divided into 24 time slots of 10ms and twelve full duplex accesses per carrier for a total of 120 possible combinations A DECT base station an RFP Radio Fixed Part can transmit all 12 possible accesses time slots simultaneously by using different frequencies or using only one frequency All signaling information 15 transmitted from the RFP within a multiframe 16 frames Voice signals are digitally encoded into a 32 kbit s signal using Adaptive Differential Pulse Code Modulation DNS Short for Domain Name System or Service or Server an Internet service that translates domain names into IP addresses DID Direct Inward Dialing Direct Inward Dialing The ability for an outside caller to dial to a PBX extension without going through an attendant or auto attendant DSP Digital Signal Processing Using computers to process signals such as sound video and other analog signals which have been converted to digital form Digital Signal Processor A specialized CPU used for digital signal processing Grandstream products all have DSP chips built inside DTMF Dual Tone Multi Frequency The standard tone pairs used on telephone terminals for dialing using i
11. size Grandstream Networks Inc 987654921 can be identical to or different from SIP User ID purposely not displayed for security protection John Doe 2 optional e g John Doe U No E yes U No E yes No ve U No E ve P minutes default 1 hour max 45 days 5062 default 5062 5008 1024 65535 default 5008 9 s Yes 101 La in audio 12 via RTP RFC2833 via SIP INFO Yes Flash will be sent DIME event if set to Yes current setting is PCMU choice 1 current setting is PCMA choice 2 current setting is G723 choice 3 current setting is G729 choice 4 current setting is G726 32 w choice 5 current setting is iLBC choice 6 up to 10 20 32 64 for G711 G726 G723 other codecs respectively 9 6 3kbps encoding rate L 5 3kbps encoding rate 9 20115 s 30ms 32 HandyTone 488 User Manual Grandstream Networks Inc payload type 97 between 96 and 127 default is 97 Silence Suppression 5 No L Yes Fax Mode E T 38 Auto Detect Pass Through Early Dial 9 L Yes use Yes only if proxy supports 484 response Dial Plan Prefix this prefix string 1s added to each dialed number Use as Dial Key No Yes if set to Yes will function as the Re Dial cy PSTN AC Termination 320 Ohm 1050 Ohm 230 nF E impedance PSTN Disco
12. the next menu option 10 HandyTone 488 User Manual Grandstream Networks Inc e returns to the main menu e 9 functions as the ENTER key in many cases to confirm an option All entered digit sequences have known lengths 2 digits for menu option and 12 digits for IP address Once all of the digits are collected they will be processed e For IP address input ignore the dot and key in the digit directly add 0 before octet with less than three digits e g IP 192 168 1 10 key in 192 168 001 010 e Key entry can not be deleted but the phone may prompt error once it is detected 5 2 Make Phone Calls 5 2 1 Calling phone or extension numbers via VoIP FXS port There are currently two methods to make an extension number call a Dial the numbers directly and wait for 4 default seconds b Dial the numbers directly and press assuming that use as dial key is selected in web configuration Examples To dial another extension on the same proxy such as 1008 simply pick up attached phone dial 1008 and then press the or wait for 4 seconds To dial a PSTN number such as 6266667890 you might need to enter in some prefix number followed by the phone number Please check with your VoIP service provider to get the information If you phone 15 assigned with a PSTN like number such as 6265556789 most likely you just follow the rule to dial 16266667890 as 1f you were calling from a regular analog phone followed by p
13. 0 5062 then the dialing convention would be Voice Prompt with option 47 then 192168001020 45062 followed by pressing the key if it is configured as a send key or wait for 4 seconds 5 2 3 Call hold This function is applicable on FXS port for VoIP calls only While in conversation pressing the FLASH button on the phone will put the remote end on hold Pressing the FLASH button again will release the previously Hold state and the bi directional media resume 5 2 4 Call waiting This function is applicable on FXS port for VoIP calls only If call waiting feature is enabled while the user is in a conversation he will hear a special stutter tone if there 1s another incoming call User can press the flash button to put the current call party on hold and switch to the other call Pressing FLASH button essentially becomes toggling between two active calls 5 2 5 Blind Transfer This function is applicable on FXS port for VoIP calls only Assume that call party A and B are in conversation A wants to Blind Transfer B to C 1 A press FLASH on the analog phone to hear the dial tone 2 Then A dials 87 then dials C s number and then or wait for 4 seconds 3 Acan hang up NOTES Enable Call Feature has to be set to YES in web configuration page Send flash event needs to be set to NO A can hold on to the phone and wait one of the three following behaviors e quick confirmation tone temporarily using the
14. 4 odd 15 9 10 a M E 15 Dd EE 15 22 16 XS VED EIGHT PATTERNGINDICA TION 16 CONFIGURATION GUIDE 18 6 1 CONFIGURING HANDYTONE 488 WAN IP THROUGH VOICE 18 PHCP MOLO 18 eate t 18 OI 18 6 2 CONFIGURING HANDYTONE 488 WITH WEB BROWSER sese 18 6 2 1 Access the Web Configuration eene 18 O22 LdU Con IP alto pelea Uus 19 0 2 3 Advanced User Configuration aeo uero dee tesa 24 General settings have same meaning as explained in above section for FXS port page Special settings on FXO port are explained below eese nnne 33 6 2 4 Saving the Configuration Changes eese nennen nennen nnns 34 6 2 5 Rebooting the 488 from Remote esses 34 2 HandyTone 488 User Manual Grandstream Networks Inc 6 3 CONFIGURATION THROUGH A CENTRAL 0020000 0000 35 7 SOR TW ARE UPGRADE 36 7 1 FIRMWARE UPGRADE THROUGH TFTP HT TP eere 36 7 2 CONFIGURAT
15. ION FILE DOWNLOAD 02 0 0000000000000000 0 reste essetis 37 7 3 FIRMWARE AND CONFIGURATION FILE PREFIX AND POSTFIX rene 37 7 4 MANAGING FIRMWARE AND CONFIGURATION FILE DOWNLOAD ce 37 8 RESTORE FACTORY DEFAULT SETTING ee ee ee ee eee eee eee eee eee eee eese 38 9 GLOSSARY OF ee eeu eve bad 39 HandyTone 488 User Manual Grandstream Networks Inc 1 Welcome Congratulations on becoming an owner of HandyTone 488 You made an excellent choice and we hope you enjoy all of its capabilities Grandstream s HandyTone 488 is an all in one VoIP integrated access device that features superb audio quality rich functionalities high level of integration compactness and ultra affordability The HandyTone 488 1s fully compatible with SIP industry standard and can interoperate with many other SIP compliant devices and software on the market Grandstream HandyTone 488 is a new addition to the popular HandyTone product family It is an enhanced model compared to the award winning HandyTone 486 in that it allows call origination and termination from to the PSTN network via FXO port remotely and automated emergency call routing through PSTN network Grandstream HandyTone 488 has been awarded the Best of Show product in 2005 Internet Telephony Conference and Expo CONFERENCE amp EXPO HandyTone 488 User Manual Grandstream Networks Inc 2 Insta
16. IPv4 to map IP network addresses to the hardware addresses used by a data link protocol The protocol operates below the network layer as a part of the interface between the OSI network and OSI link layer It is used when IPv4 is used over Ethernet ATA Analogue Telephone Adapter Covert analogue telephone to be used in data network for VoIP like Grandstream HT series products CODEC Abbreviation for Coder Decoder It s analog to digital A D and digital to analog D A converter for translating the signals from the outside world to digital and back again CNG Comfort Noise Generator geneate artificial background noise used in radio and wireless communications to fill the silent time in a transmission resulting from voice activity detection DATAGRAM A data packet carrying its own address information so it can be independently routed from its source to the destination computer DECIMATE To discard portions of a signal in order to reduce the amount of information to be encoded or compressed Lossy compression algorithms ordinarily decimate while subsampling DECT Digital Enhanced Cordless Telecommunications A standard developed by the European Telecommunication Standard Institute from 1988 governing pan European digital mobile telephony DECT covers wireless PBXs telepoint residential cordless telephones wireless access to the public switched telephone network Closed User Groups CUGs Local Area Networks and wireless local
17. Tag 802 1p priority value 0 7 No Key Entry Timeout NAT Traversal seconds default is 4 seconds Li No Yes STUN server 15 URI or IP port keep alive interval in seconds default 20 seconds Use NAT IP used in SIP SDP message if specified Firmware Upgrade Upgrade Via fm grandstream com gs Firmware Server Path fm grandstream com gs Config Server Path Firmware File Prefix Firmware File Postfix Config File Prefix Config File Postfix Automatic Upgrade 10080 L Yes check for upgrade every minutes default 7 days 24 HandyTone 488 User Manual Grandstream Networks Inc 9 Always Check for New Firmware L Check New Firmware only when F W pre suffix changes Firmware Key OO in Hexadecimal Representation NTP Server tmeristgov URI or IP address WAN side http access No 9 Yes WAN side access http server will be rejected if set to No Admin Password Layer 3 QoS Layer 2 QoS No Key Entry Timeout NAT Traversal Keep alive interval Use NAT IP Firmware Upgrade and Provisioning Administrator password Only administrator can configure the Advanced Settings FXS port and FXO port page Password field is purposely blanked for security reason after clicking UPDATE button The maximum password length is 25 characters This field defines the layer
18. User Manual Handy 488 Analog Telephone Adaptor For SW Release Version 1 0 3 18 Grandstream Networks Inc www grandstream com G NETWORKS ndstrean nnovative IP Telephony HandyTone 488 User Manual Grandstream Networks Inc Table of Contents L WELCOME 4 2 MINS AAA TION 5 3 WHAT IS INCLUDED IN THE 2 7 DJ QOPREDPCOMPEDONGBS nea oie tata tae 7 SP WARRANTY MEET CO 7 4 PRODUCI OVERVIEW e NRI Eein 8 4o WOE YR BATU REG oho hat coc va uten he oa LB E 8 4 2 HARDWARE S PECIRIGATION 2223 99 0299 120 E 9 3 BASIC OPERA TIONS 10 3 GET PAMILIAR WITH bee das 10 2222 MAKE PRONE C 11 5 2 1 Calling phone or extension numbers via VoIP FXS 11 220 Dre IP eall ed un eue 11 2 2 5 Call hold uisus iste dae 12 E MESSER 72 12 320 ette er 13 ODE o Dac 15 DO PSTN PASS petu ee 15 DF SIN 13 JS T0 PSPNSIOSVOLPC 1
19. a but does not transport any data itself It is used periodically to transmit control packets to participants in a streaming multimedia session The primary function of RTCP is to provide feedback on the quality of service being provided by RTP Real time Transport Protocol defines a standardized packet format for delivering audio and video over the Internet It was developed by the Audio Video Transport Working Group of the and first published in 1996 as RFC 1889 Session Description Protocol is a format for describing streaming media initialization parameters It has been published by the IETF as RFC 2327 Session Initiation Protocol An IP telephony signaling protocol developed by the IETF RFC3261 SIP is a text based protocol suitable for integrated voice data applications SIP is designed for voice transmission and uses fewer resources and is considerably less complex than H 323 Grandstream products are SIP based Simple Traversal of UDP over is a network protocol allowing clients behind NAT or multiple NATs to find out its public address the type of NAT it is behind and the internet side port associated by the NAT with a particular local port This information 1s used to set up UDP communication between two hosts that are both behind NAT routers The protocol is defined in REC 3489 STUN will usually work good with non symmetric NAT routers Transmission Control Protocol is one of the core protocols of the Inter
20. a No o Yes Unregister On Reboot No Yes Register Expiration is in minutes default 1 hour max 45 days local SIP port ck default 5060 local RTP port 594 1024 65535 default 5004 Use random port E No Was DTMF Payload 101 Type Send in audio via RFC2833 via SIP INFO Send Flash Event No L Yes Flash will be sent as a DTMF event if set to Yes Enable Call E 9 Yes Yes Call Forwarding amp Call Waiting Disable are Features supported locally Offhook Auto Dial User ID extension to dial automatically when offhook Proxy Require Disable Call Waiting No Yes Preferred Vocoder current setting is PCMU EN choice 1 in listed order current setting is POMA choice 2 current setting is G723 3 current setting is G729 choice 4 current setting is 2726 32 choice 5 current setting is iLBC choice 6 Voice Frames per TX up to 10 20 32 64 for G711 G726 G723 other codecs respectively G723 rate Ei 6 3kbps encoding rate L 5 3kbps encoding rate frame size V5 205 E 30ms payload type between 96 and 127 default is 97 Silence Suppression o No Yes 27 HandyTone 488 User Manual Grandstream Networks Inc Fax Mode E T 38 Auto Detect Pass Through Early Dial 9 L Yes use Yes only if proxy supports 484 r
21. a Web Interface as well as via Configuration File through TFTP or HTTP Config Server Path is the TFTP or HTTP server path for configuration file It needs to be set to a valid URL either in FQDN or IP address format The Config Server Path can be same or different from the Firmware Server Path A configuration parameter is associated with each particular field in the web configuration page A parameter consists of a Capital letter P and 2 to 3 Could be extended to 4 in the future digit numeric numbers 1 P2 1s 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 C gXXXXXXXXXXxX Where 15 the address of the device 1 cfg000b820102ab The configuration file name should be in lower cases 7 3 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 the possible to store ALL of the firmware with different version in one single directory Similarly Config File Prefix and Postfix allows device to download the configuration file with the matching Prefix and Postfix Thus multiple configuration files for the same device can be stored in one directory In addit
22. ade and Provisioning upgrade via field needs to be set to or HTTP respectively Firmware Server Path needs to be set to a valid URL of a HTTP server server name can be in either FQDN or IP address format are examples of some valid URL e g firmware mycompany com 6688 Grandstream 1 0 3 18 e g 168 75 215 189 NOTES TFTP server in IP address format can be configured via IVR Please refer to section 6 1 3 for instructions If TFTP server is in FQDN format it must be set via web configuration interface e 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 15 available the HandyTone ATA will attempt to retrieve the new image files by downloading them into the HandyTone ATA s SRAM During this stage the HandyTone ATA 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 If fails for any reason e g server 15 not responding there are no code image files available for upgrade or checksum test fails etc the HandyTone ATA will stop the TFTP HTTP process and simply boot using the existing code image in the flash e Firmware upgrade may take as long as 1 to 20 minutes over Internet or just 20 seconds if it 1 performed on a LAN It is recommended to conduct f
23. ber located on the bottom of the unit Step 2 Encode the MAC address Please use the following mapping 0 9 0 9 A 22 B 222 C 2222 D 33 E 333 F 3333 For example if the MAC address 1s 000b8200e395 it should be encoded as 0002228200333395 Step 3 To perform factory reset Press or the LED button for voice prompt Enter 99 and get the voice prompt Reset Enter the encoded MAC address of the device Wait for 15 seconds device will reboot automatically and restore to factory default setting NOTES e Please be aware by default the HandyTone 488 WAN side HTTP access is disabled After a factory reset the device s web configuration page can be accessed only from its LAN port please refer to instructions in section 6 2 1 for details 38 HandyTone 488 User Manual Grandstream Networks Inc 9 Glossary of Terms ADSL Asymmetric Digital Subscriber Line Modems attached to twisted pair copper wiring that transmit from 1 5 Mbps to 9 Mbps downstream to the subscriber and from 16 kbps to 800 kbps upstream depending on line distance AGC Automatic Gain Control is an electronic system found in many types of devices Its purpose is to control the gain of a system in order to maintain some measure of performance over a changing range of real world conditions ARP Address Resolution Protocol is a protocol used by the Internet Protocol 826 pecifically
24. call waiting indication tone follows by a dial tone This indicates the transfer has been successful e A quick busy tone followed by a restored call On supported platforms only This means that is transfered to C with no success The busy tone is just to indicate A that the transfer has failed e Busy tone keeps playing This means A have failed to receive the second NOTIFY from B and decide to time out Note this does not indicate the transfer has been successful nor does it indicate the transfer has failed When B 1 a client that does not support the second NOTIFY this will be the case In bad network scenarios this could also happen although the transfer may have been completed successfully 12 HandyTone 488 User Manual Grandstream Networks Inc 5 2 6 Attended Transfer This function is applicable on FXS port for VoIP calls only Assuming that call party A and B are in conversation A wants to Attend Transfer B to C 1 A presses FLASH on the analog phone to get a dial tone 2 A then dial C s number followed by or wait for 4 seconds 3 If C answers the call and are in conversation Then A hang up to complete transfer 4 If C does not answer the call A can press FLASH back to talk to B NOTES e When antended Transfer failed and A hang up the HandyTone 488 will ring user back again to remind that is still on the call A can pick up the phone to restore conversation with 5 2 7 3 way Confe
25. ck 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 1 configured or the dial tone otherwise Enter the pin code that is configurable on the configuration page The caller will hear the dial tone get connected to the PSTN line if the pin code 15 valid otherwise the continuous tone is played again to prompt caller to enter in the pin code again The use may try up to 3 times to enter pin code if none is valid HT488 will hang up After the caller hears dial tone from PSTN line the caller can start dialing number to make calls Users can choose whether apply password protection for VoIP to PSTN calls or not A PIN Pin for PSTN calls consists of up to 8 numeric digits can be configured through BASIC SETTINGS of the web configuration page By default there is no password protection 1 e there is no authentication required for callers on the use of PSTN line through HT488 When a PIN is configured for VOIP to PSTN call flow the VoIP device that calls into the HT488 FXO account needs to configure RFC2833 or SIP Info for digit transmission Upon hearing the special continuous tone for PIN code input if the caller don t enter any digit HT488 will time out and hang up the call in 10 seconds During any stage of digits input a 4 seconds timeout is applied to serve as an end of PIN or destination number input Users may also u
26. ect the other end of the Ethernet cable to an uplink port a router or a modem etc Connect a PC to the LAN port of HandyTone 488 if HT488 is used as a router 5 Insert the power adapter into the HandyTone 488 and connect it to a wall outlet Please follow the instructions in section 6 2 to configure the HandyTone 488 HandyTone 488 User Manual Grandstream Networks Inc 3 What is Included the Package The HandyTone 488 package contains 1 One HandyTone 488 2 One universal power adaptor 3 One Ethernet cable 3 1 Safety Compliances The HandyTone 488 is compliant with various safety standards including FCC CE and C tick Its power adaptor is compliant with UL standard The HandyTone 488 should only operate with the universal power adaptor provided in the package 3 2 Warranty Grandstream has a reseller agreement with our reseller customer End users should contact the company from whom you purchased the product for replacement repair or refund If you purchased the product directly from Grandstream contact your Grandstream Sales and Service Representative for a RMA Return Materials Authorization number Grandstream reserves the right to remedy warranty policy without prior notification Warning Please do not attempt to use a different power adaptor Using other power adaptor may damage the HandyTone 488 and will void the manufacturer warranty Caution Changes or modifications to this product not expressly appr
27. en saved reboot configuranon changes They wil take effect on next Reboot 6 2 5 Rebooting the HandyTone 488 from Remote User can then power cycle the device or reboot HandyTone ATA by clicking on the Reboot button at the bottom of the configuration page Once done the following screen will be displayed to indicate that rebooting is underway The device 15 rebooting now You may relogin by clicking on the link below in 30 seconds Click to relogin Note e DONOT INTERUPT THE BOOTING UP PROCESS OR THE DEVICE COULD BE DAMAGED 34 HandyTone 488 User Manual Grandstream Networks Inc 6 3 Configuration through a Central Server Grandstream HandyTone ATAs can be automatically configured from a central provisioning system When HandyTone ATA boot up it will send or request to download configuration file cfg000bS2xxxxxx where 000582 is the MAC address of the HandyTone ATA The configuration files be downloaded via or HTTP from the central server A service provider or an enterprise with large deployment of HandyTone ATA can easily manage the configuration and service provisioning of individual devices remotely from a central server Grandstream provides a licensed provisioning system called GAPS that can be used to support automated configuration of HandyTone ATA GAPS Grandstream Automated Provisioning System uses enhanced NAT friendly TFTP or
28. esponse Dial Plan Prefix this prefix string is added to each dialed number Use as Dial Key 9 Yes if set to Yes will function as the Re Dial ey E No do not send SUBSCRIBE for Message Waiting Indication s Yes send periodical SUBSCRIBE for Message Waiting Indication SUBSCRIBE for MWI Send Anonymous No 9 Yes caller ID will be blocked if set to Yes 9 Yes Special Feature Standard FXS Impedance 600 Ohm North America Caller ID Scheme Bellcore North America Onhook Voltage 36V Lock keypad update L Yes configuration update via keypad is disabled 1f set to 9 Yes reverse polarity upon call establishment and Polarity Reversal termination SIP Server SIP Server s Domain name or IP address provided by VoIP service provider Outbound Proxy 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 15 detected STUN will not work and ONLY outbound proxy will provide solution for it This information is provided by VoIP service provider SIP User ID User account information provided by VoIP service provider ITSP usually has the form of digit similar to phone number or actually a phone number Authenticate ID ID used for authentication usually same as SIP user ID but could be different and decided by ITSP A
29. ess of the ATA if you use this feature as it will prevent you to access the IVR and the only way to access the device configuration is via the web configuration page SIP Extension to notify SIP server that the unit is behind the NAT Firewall Default is No The HandyTone ATA supports 6 different Vocoder types including G 711 A U law G 723 1 G 726 32 G 729A iLBC Users can configure Vocoders in a preference list that will be included with the same preference order in SDP message 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 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 e g if the first vocoder is configured as G723 and the Voice Frames per TX is set to be 2 then the ptime value in the SDP message of an INVITE request will be 60ms because each G723 voice frame contains 30ms of audio similarly if this field 1s set to be 2 and if the first vocoder chosen 15 G729 or G711 G726 then the value in the SDP message of an INVITE request will be 20ms If the configured voice frames per TX exceeds the maximum allowed value the HandyTone ATA will use and save the maximum allowed value for the corresponding f
30. eters such as the subnet mask and default router and to provide other configuration information such as the addresses for printer time and news servers ECHO CANCELLATION Echo Cancellation 15 used in telephony to describe the process of removing echo from a voice communication in order to improve voice quality on a telephone call In addition to improving quality this process improves bandwidth savings achieved through silence suppression by preventing echo from traveling across a network There are two types of echo of relevance in telephony acoustic echo and hybrid echo Speech compression techniques and digital processing delay often contribute to echo generation in telephone networks H 323 A suite of standards for multimedia conferences on traditional packet switched networks HTTP Hyper Text Transfer Protocol the World Wide Web protocol that performs the request and retrieve functions of a server IP Internet Protocol A packet based protocol for delivering data across networks IP PBX IP based Private Branch Exchange IP Telephony Internet Protocol telephony also known as Voice over IP Telephony A general term for the technologies that use the Internet Protocol s packet switched connections to exchange voice fax and other forms of information that have traditionally been carried over the dedicated circuit switched connections of the public switched telephone network PSTN The basic steps involved in originating an IP
31. ion when the field Check New Firmware only when F W pre suffix changes is set to Yes the device will only issue firmware upgrade request if there are changes the firmware Prefix or Postfix 7 4 Managing Firmware and Configuration File Download When Automatic Upgrade is set to Yes Service Provider can use P193 Auto Check Interval in minutes default and minimum is 60 minutes to have the devices periodically check with either Firmware Server or Config Server whenever they are defined This allows the device periodically check if there are any new changes need to be taken on a scheduled time By defining different intervals in P193 for different devices Server Provider can spread the Firmware or Configuration File download in minutes to reduce the Firmware or Provisioning Server load at any given time 37 HandyTone 488 User Manual Grandstream Networks Inc 8 Restore Factory Default Setting Warning Restore the Factory Default Setting will DELETE all configuration information of the device 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 service provider Please disconnect network cable and power cycle the unit before trying to reset the unit to factory default The steps are as follows Step 1 Find the MAC Address of the device It is a 12 digits HEX num
32. irmware upgrade in a controlled LAN environment if possible For users who do not have a local firmware upgrade server Grandstream provides a NAT friendly server on the public Internet for firmware upgrade Please check the Services section of Grandstream s Web site to obtain our public server s IP address e Alternatively user can download a free or HTTP server and conduct local firmware upgrade free windows version TFTP server is available for download from http support solarwinds net updates New customerFree cfm Our latest official release can be downloaded from http www grandstream com y firmware htm Unzip the file and put all of them under the root directory of the TFTP server Put the PC running the TFTP server and the HandyTone ATA in the same LAN segment 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 Start the server in the HandyTone ATA s web configuration page configure the Firmware Server Path with the IP address of the PC update the change and reboot the unit Please be advised that our client will pull out firmware from the WAN side if 36 HandyTone 488 User Manual Grandstream Networks Inc the TFTP server is connected to the device s LAN port the firmware upgrade will not work by design 7 2 Configuration File Download Grandstream SIP Device can be configured vi
33. irst vocoder choice The maximum value for PCM 15 10 x10ms frames for G726 it is 20 x10ms frames for G723 it is 32 x30ms frames for G729 G728 64 x10ms and 64 x2 5ms frames respectively This defines the encoding rate for G723 vocoder Default setting 1s 6 3kbps This sets the 1L BC size 20ms 30ms This defines payload type for iLBC Default value is 97 The valid range 15 between 96 and 127 This controls the silence suppression V AD feature of G723 and G729 If set to Yes when a silence is detected small quantity of VAD packets instead of audio packets will be sent during the period of no talking If set to No this feature 1s disabled T 38 Auto Detect FoIP by default or fax Pass Through 30 HandyTone 488 User Manual Grandstream Networks Inc Early Dial Default is No Use only if proxy supports 484 response Dial Plan Prefix Sets the prefix added to each dialed number Use as This parameter allows users to configure the 4 key to be used as the Send Send Key or Dial key If set to Yes pressing this key will immediately trigger the sending of dialed string collected so far If set to this key will then be included as part of the dial string to be sent out Subscribe for MWI Default is No When set to Yes a SUBSCRIBE for Message Waiting Indication will be sent periodically Send Anonymous If this parameter is set to Yes user ID will be sent as
34. is very important ID for ISP troubleshooting This field shows WAN port IP address This field contains the product model info such as HT488 Program This is the main software release This number is always used for firmware upgrade Current release is 1 0 3 18 Bootloader current version is 1 0 8 9 HTML current version 1 0 3 18 VOC current version is 1 0 0 10 This shows system up time since last reboot Whether the unit is registered to the SIP server This shows whether the PPPoE link is up if connected to DSL modem This shows what kind of NAT the HandyTone is behind NAT detection mechanism is based on STUN protocol Basic settings page 20 HandyTone 488 User Manual Grandstream Networks Inc End User Password purposely not displayed for security protection IP Address 9 dynamically assigned via DHCP default or PPPoE will attempt PPPoE if DHCP fails and following is non blank PPPoE account ID o password PPPoE Service Name Do Preferred DNS server 9 L statically configured as IP Address 192 192 160 Subnet Mask fo Default Router fo Jo DNS Server 1 fo DNS Server 2 fo Jo Time Zone GMT 5 00 US Eastern Time New York Daylight 1 No L Yes if set to Yes display time will be 1 hour ahead of normal Savings Time time NAT DHCP Server Information amp Configuration Cloned WAN MAC Addr m
35. ll be initiated from PSTN line For ex if Route call to PSTN is configured to be 626 all outgoing calls start with 626 will be initiated from PSTN line 5 3 Call Features All the call feature codes are applicable to FXS port for VoIP calls only 5 3 1 Features Table Following table shows the call features of HandyTone 488 70 TA Unconditional Call Forward To use this feature dial 72 and get the dial tone Then dial 15 HandyTone 488 User Manual Grandstream Networks Inc the forward number and for dial tone then hang up 73 Cancel Unconditional Call Forward To cancel Unconditional Call Forward dial 73 and get the dial tone then hang up 90 Busy Call Forward To use this feature dial 90 and get the dial tone Then dial the forward number and for a dial tone then hang up 9 Cancel Busy Call Forward To cancel Busy Call Forward dial 91 and get the dial tone then hang up 7 Delayed Call Forward To use this feature dial 92 and get the dial tone Then dial the forward number and for a dial tone then hang up 93 Cancel Delayed Call Forward To cancel this Forward dial 93 and get the dial tone then hang up Flash Hook When conversation this action will switch to the new incoming call if there 1 a call waiting indication When in conversation without an incoming call this action will switch
36. llation HandyTone 488 Analog Telephone Adaptor is an all in one VoIP integrated device designed to be a total solution for networks providing VoIP services The HandyTone 488 VoIP functionalities are available via a regular analog telephone The following photo illustrates the appearance of a HandyTone 488 Top View Side Views RJ45 10M Ethernet FXS Port LAN WAN Phone 1 4 5 1200 mE ua TE ME ad BUTTON FXO Port RED LED Phone Line GREEN LED Interconnection Diagram of the HandyTone 488 Internet ADSL Cable Modem Ethernet Analog Phone PSTN Cordless Phone LAN PC PC PC Fax HandyTone 488 User Manual Grandstream Networks Inc HandyTone 488 has one FXS port and one FXO port The PHONE port next to the LAN port is a FXS port The LINE port on the side of the HandyTone 488 is a FXO port Both the FXS port and the FXO port can have a separate SIP account This is a key feature of HandyTone 488 as it supports simultaneous call on both FXS port and FXO port Telephone calls can be originated from or terminated on the PSTN network via FXO port remotely Following are the steps to install a HandyTone 488 1 Connect a standard touch tone analog telephone to the PHONE port 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 HandyTone 488 and conn
37. lue is 120hr 5 Days The time address are assigned to the LAN clients Forward all WAN IP traffic to a specific IP address no matching port is used by HandyTone 488 itself or in the defined port forwarding Allow users to forward a matching TCP UDP port to a specific LAN IP address with a specific TCP UDP port Default 15 4 It specifies number of phone rings before a PSTN incoming call is bridged to VoIP The code to access the PSTN line Default is 00 PIN code to bridge from VoIP to PSTN PIN code to bridge from PSTN to VoIP If the dialed digits match one of the specified prefix here outbound calls will be initiated from PSTN line This field is especially useful for emergency calls 23 HandyTone 488 User Manual Grandstream Networks Inc Forward to PSTN Calls are unconditionally forwarded to the specified PSTN phone number for all incoming VoIP calls on FXO port Forward to VoIP Calls are unconditionally forwarded to the specified VoIP phone number for all incoming PSTN calls 6 2 3 Advanced User Configuration To login to the Advanced User Configuration page please follow the instructions in section 6 2 1 to get to the following login page The password is case sensitive and the factory default password for Advanced User is admin Advanced settings Admin Password purposely not displayed for security protection Layer 3 Qos Diff Serv or Precedence value Layer 2 QoS 802 1Q VLAN
38. m m m m in hex format LAN Subnet 255 255 2550 9497999 default is 255 255 255 0 LAN DHCP 19246821 0 Base IP sibi base IP for the LAN port default is 192 168 2 1 DHCP IP Tog no in units of hours default is 120 hours 5 days DMZ IP 0 0 UDP Onl Port wan port LAN IP LAN port Protocol 21 HandyTone 488 User Manual Grandstream Networks Inc Forwarding xd p m EN sj LAN IP LAN port EN Protocol Pony WAN port o LAN IP LAN port po Protocol UDPOnly MANDEN i LAN IP LAN port Protocol UDPOnly LAN IP LAN port Protocol UDPOnly 0 0 UDP WAN port LAN IP LAN port Protocol y 0 0 UDP Onl WAN port LAN IP LAN port Protocol kd Number of number of phone rings before a PSTN incoming call 1 forwarded default Rings 4 PSTN access mE code key pattern to use PSTN line default is 00 PIN for PSTN Enter digits to authorize calling PSTN numbers from VOIP no Calls default PIN for VOIP Enter digits to authorize calling VOIP terminals from PSTN no Calls default Outbound calls will be routed to PSTN port when dialed digits match one of the following Route Call to PSTN 50 Forward to VoIP calls will be forwarded to the specified PSTN PSTN number if ring no answer Forward to PSTN calls will be forwarded to the specified VoIP VoIP numbe
39. n band signaling The standards define 16 tone pairs 0 9 and A F although most terminals support only 12 of them 0 9 and FQDN Fully Qualified Domain Name A FQDN consists of a host and domain name including top level domain For example www grandstream com is a fully qualified domain name www is the host grandstream is the second level domain and com is the top level domain FXO Foreign eXchange Office An device can be an analog phone answering machine fax or anything that handles a call from the telephone company like AT amp T They should also operate the same way when connected to an FXS interface An FXO interface will accept calls from FXS or PSTN interfaces All countries and regions have their own standards FXO is complimentary to FXS and the PSTN 40 HandyTone 488 User Manual Grandstream Networks Inc FXS Foreign eXchange Station An FXS device has hardware to generate the ring signal to the FXO extension usually an analog phone An FXS device will allow any FXO device to operate as if it were connected to the phone company This makes your PBX the POTS PSTN for the phone The FXS Interface connects to FXO devices by an FXO interface of course DHCP The Dynamic Host Configuration Protocol DHCP is an Internet protocol for automating the configuration of computers that use TCP IP DHCP can be used to automatically assign IP addresses to deliver TCP IP stack configuration param
40. n server Firmware File Default is blank If configured HT488 will request the firmware file Prefix with the prefix This setting 15 useful for ITSPs End user should keep it blank Firmware File Default is blank End user should keep it blank Postfix Config File Prefix Default is blank End user should keep it blank Config File Postfix Default is blank End user should keep it blank Automatic Upgrade Default is Yes Firmware Key For firmware encryption It should be 32 digit in Hexadecimal Representation End user should keep it blank NTP server URI or IP address of the NTP Network Time Protocol server which the HandyTone ATA will use to synchronize the date time WAN side http Default is The access to configuration page via WAN port is disabled access Need to change to Yes if user wants WAN side HTTP access to the ATA FXS port page E 1NGS EI NOS SIP Server e g sip mycompany com IP address Outbound Proxy e g proxy myprovider com or IP address if any SIP User ID didi the user part of an SIP address Authenticate ID be identical to or different from SIP User ID Authenticate purposely not displayed for security protection Password Name wel optional e g John Doe Use DNS SRV E User ID is phone o No Yes HandyTone 488 User Manual Grandstream Networks Inc number SIP Registration E
41. net protocol suite Using TCP applications on networked hosts can create connections to one another over which they can exchange data or packets The protocol guarantees reliable and in order delivery of sender to receiver data Trivial File Transfer Protocol is a very simple file transfer protocol with the functionality of a very basic form of FTP It uses UDP port 69 as its transport protocol User Datagram Protocol UDP is one of the core protocols of the Internet protocol suite Using UDP programs on networked computers can send short messages known as datagrams to one 43 HandyTone 488 User Manual Grandstream Networks Inc another UDP does not provide the reliability and ordering guarantees that does datagrams may arrive out of order or go missing without notice However as a result UDP is faster and more efficient for many lightweight or time sensitive purposes VAD Voice Activity Detection or Voice Activity Detector is an algorithm used in speech processing wherein the presence or absence of human speech is detected from the audio samples VLAN A virtual LAN known as a VLAN is a logically independent network Several VLANs co exist on a single physical switch It is usually refer to the IEEE 802 1Q tagging protocol VoIP Voice over VoIP encompasses many protocols All the protocols do some form of signalling of call capabilities and transport of voice data from one point to another e g SIP
42. nnect Tone Frequency 399 f2 iai Hz 0 inactive default 1s 480Hz 620Hz PSTN Disconnect Tone Cadence Choice 1 Off 0 disabled Choice 2 On off ms 0 disabled Choice 3 On off ms 0 disabled PSTN Silence Timeout sec terminate call after long silence detected default is 60 sec max 65536 Update General settings have same meaning as explained in above section for FXS port page Special settings on FXO port are explained below Local SIP port The default value for FXO port is 5062 Local RTP port The default value for FXO port is 5008 PSTN AC Selects the impedance of the analog PSTN line connected to the Line port Termination PSTN Disconnect This configuration should be configured by the VoIP service provider Some Tone country use single frequency tone to signal PSTN disconnection some country use double frequency tone PSTN Disconnect This setting can be configured to suit the telephone company s standard in Tone Cadence different country 33 HandyTone 488 User Manual Grandstream Networks Inc PSTN Silence Terminate call after long silence detected Default setting 15 60 sec max 65536 Timeout 6 2 4 Saving the Configuration Changes Once a change is made users should click on the Update button on the Configuration page The HandyTone ATA will then display the following screen to confirm that the changes have been saved have be
43. on or dial 6 amp 7 from the analog phone The following table shows how to use the voice prompt menu to configure the device Main Menu Enter a Menu Option Enter for the next menu option Enter to return to the main menu Enter 01 06 47 86 99 menu option DHCP Mode Enter 9 to toggle the selection Static IP Mode If user selects Static IP Mode user need configure all the IP address information through menu 02 to 05 If user selects Dynamic IP Mode the device will retrieve all IP address information from DHCP server automatically when user reboots the device Address IP address The current WAN IP address 1 announced Enter 12 digit new IP address if in Static IP Mode be 08 yt 6 DNS Server IP address TETP Server IP address 4 Direct IP Calling When entered you will be prompted a dial tone then enter 12 digit IP address This menu can also be entered by pressing the button again For details see 4 2 2 Make a Direct Call RESET Enter 9 to reboot the phone Enter encoded MAC address to restore factory default setting Invalid Entry Automatically returns to main menu 01 02 03 04 05 7 e Once the button is pressed it enters the voice prompt main menu If the button is pressed again while it is already in the voice prompt menu it jumps to Direct IP Call option and a dial tone is prompted e shifts down to
44. oved by Grandstream or operation of this product in any way other than as detailed by this User Manual could void your manufacturer warranty Information in this document is subject to change without notice No part of this document may be reproduced or transmitted in any form or by any means electronic or mechanical for any purpose without the express written permission of Grandstream Networks Inc HandyTone 488 User Manual Grandstream Networks Inc 4 1 4 Product Overview Key Features Supports SIP 2 3261 TCP UDP IP RTP RTCP HTTP ICMP ARP RARP DNS DHCP both client and server NTP PPPoE STUN etc Built in router NAT Gateway and DMZ port forwarding Supports call origination and termination from to the PSTN network via FXO port Powerful digital signal processing DSP to ensure superb audio quality advanced adaptive jitter control and packet loss concealment technology Support various vocoders including G 711 PCM a law and u law G 723 1 5 3K 6 3K G 726 32K as well as G 729A and 1LBC Support advanced call features on FXS port includes Caller ID Name display or block Hold Call Waiting Flash Call Transfer Call Forward 3 way conference in band and out of band DTMF etc Support fax pass through for PCMU and and T 38 FoIP Fax over IP Support Silence Suppression VAD Voice Activity Detection CNG Comfort Noise Generation Line Echo Cancellation G 168 and AGC
45. port used is UDP 123 Grandstream products using NTP to get time from Internet OBP SBC PPPoE PSTN Outbound Proxy or another name Session Border Controller A device used in VoIP networks OBP SBCs are put into the signaling and media path between calling and called party The OBP SBC acts as if it was the called VoIP phone and places a second call to the called party The effect of this behaviour is that not only the signaling traffic but also the media traffic voice video etc crosses the OBP SBC Without an OBP SBC the media traffic travels directly between the VoIP phones Private OBP SBCs are used along with firewalls to enable VoIP calls to and from a protected enterprise network Public VoIP service providers use OBP SBCs to allow the use of VoIP protocols from private networks with internet connections using NAT Point to Point Protocol over Ethernet is a network protocol for encapsulating PPP frames in Ethernet frames It is used mainly with cable modem and DSL services Public Switched Telephone Network 42 HandyTone 488 User Manual Grandstream Networks Inc RTCP RTP SDP SIP STUN TCP TETP UDP 1 the phone service we use for every ordinary phone call or called POT Plain Old Telephone or circuit switched network Real time Transport Control Protocol defined in RFC 3550 a sister protocol of the Real time Transport Protocol RTP It partners RTP in the delivery and packaging of multimedia dat
46. r End User This contains the password to access the Web Configuration Menu This Password field is case sensitive with max 25 characters HandyTone 488 User Manual IP Address Time Zone Daylight Savings Time Cloned WAN MAC Address LAN Subnet Mask LAN DHCP Base IP DHCP IP Lease Time DMZ IP Port Forwarding Number of rings PSTN access code PIN for PSTN calls PIN for VoIP calls Route Call to PSTN Grandstream Networks Inc This setting is for the WAN port If DHCP mode 15 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 HandyTone ATA will acquire its IP address from DHCP in the network PPPoE settings is usually for DSL ADSL modem users The HandyTone will attempt to establish a PPPoE session if PPPoE account is set If Static IP mode 15 selected the IP address Subnet Mask Default Router IP address DNS Server 1 mandatory DNS Server 2 optional fields need to be configured Displayed date time will be adjusted according to the specified time zone Default NO If set to Yes then the displayed time will be 1 hour ahead of normal time Allow the user to set a specific MAC address Set in Hex format Sets the LAN subnet mask Default value 15 255 255 255 0 Base IP for the LAN port which functions as default gateway for its LAN Default value is 192 168 2 1 Value is set in units of hours Default va
47. rencing This function is applicable on FXS port for VoIP calls only Assuming that call party and are in conversation wants to bring C in a conference A press FLASH button on the analog phone to get a dial tone A dials 23 followed by C s number and or wait for 4 seconds If C answers the call then A press FLASH to bring B C in the conference If C does not answer the call A can press FLASH back to talk to B juge ue es NOTES Enable Call Feature has to be set to YES in web configuration page 5 2 8 PSTN Pass Through life line HandyTone 488 supports PSTN pass through user can send and receive PSTN call with attached analog phone To receive PSTN calls simply make phone off hook when the analog phone rings To make a PSTN call simply press the PSTN access code 00 15 default or any number configured in web configuration page to switch to the PSTN line and get dial tone then dial the PSTN number When HandyTone 488 is out of power it will function as a jack The user will be automatically connected to the PSTN Line 5 2 9 VoIP to PSTN Calls This function is applicable on FXO port that functions as a bridge between VoIP and PSTN The user can remotely use PSTN line to initiate a call 13 HandyTone 488 User Manual Grandstream Networks Inc To make a VoIP to PSTN call 1 Note Dial the FXO SIP account phone number to establish the VoIP session The caller will hear the ring ba
48. ressing the or wait for 4 seconds 5 2 2 Direct IP calls Direct IP calling allows two parties that 15 a HandyTone with an analog phone and another VoIP Device to talk to each other in an ad hoc fashion without a SIP proxy This kind of VoIP calls can be made between two parties if e both HandyTone and other VoIP Device 1 e another HandyTone or Budgetone SIP phone or other VoIP unit have public IP addresses or e both HandyTone and other VoIP Device are on the same LAN using private IP addresses or e Both HandyTone ATA and other VoIP Device can be connected through a router using public or private IP addresses with necessary port forwarding or DMZ To make a direct IP to IP call first pick up the analog phone or turn on the speakerphone on the analog phone then follow Section 4 1 with voice prompt 47 followed by the 12 digit target IP address User will hear a voice prompt Direct IP Calling and a dial tone Enter a 12 digit target IP address to make a call Destination ports can be specified by using 4 encoding for followed by the port number Examples 11 HandyTone 488 User Manual Grandstream Networks Inc If the target IP address is 192 168 0 10 the dialing convention is Voice Prompt with option 47 then 192 168 000 010 followed by pressing the key if it is configured as a send key or wait for more than 5 seconds If the target IP address port is 192 168 1 2
49. se the 4 key to indicate the end of an input On the web configuration page if the Forward to PSTN is configured the second stage dialing is eliminated 1 e after dialing into the FXO SIP account number the PSTN number will be called automatically 5 2 10 PSTN to VoIP Calls This function is applicable on FXO port that functions as a bridge between VoIP and PSTN The user can make VoIP calls remotely by dialing into FXO Line port on HT488 To make a PSTN to VOoIP call l 2 Make an incoming call to the PSTN line on port The attached analog phone will ring for 4 times by default this setting 1s configurable on the configuration page If no one picks up the phone on FXS port after 4 rings then the caller hears either a special continuous tone or a dial tone The continuous tone is played if the pin code is configured or the dial tone otherwise Enter the pin code that 1s configurable on the configuration page The caller will hear the dial tone and get bridged to VoIP if the pin code is valid otherwise the continuous tone is played again to prompt caller to enter in the pin code again The use may try up to 3 times to enter in pin code if none 1 valid 488 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 account configured on the port 14 HandyTone 488 User Manual Grandstream Networks Inc Note
50. smit It is the base RTP port for channel 0 When configured channel 0 will use this port value for RTP and the port_value 1 for its RTCP channel 1 will use port 2 for RTP and port_value 3 for its RTCP The default value for FXS port 1s 5004 Default No If set to Yes the device will pick randomly generated SIP and RTP ports This 1s usually necessary when multiple HandyTone ATAs are behind the same NAT This parameter sets the payload type for using RFC2833 This parameter specify the mechanism to transmit DTMF digit There are 3 modes supported in audio which means DTMF is combined in audio signal not very reliable with low bit rate codec RTP 2833 or SIP INFO Default is NO If set to yes flash will be sent as DTMF event Default is Yes Advance call features and feature codes functions are supported locally 29 HandyTone 488 User Manual Grandstream Networks Inc Offhook Auto Dial Proxy Require Disable Call Waiting Preferred Vocoder Voice Frames per TX G723 Rate frame size iLBC payload type Silence Suppression Fax Mode This parameter allows users to configure a User ID or extension number to be automatically dialed upon offhook Please note that only the user part of a SIP address needs to be entered here The HandyTone ATA will automatically append the and the host portion of the corresponding SIP address Note Please write down the IP addr
51. to a new channel for a new call 5 4 Fax HandyTone 488 supports FAX in two modes T 38 Fax over and fax pass through T 38 15 the preferred method because it 1 more reliable and works well in most network conditions If the service provider supports T 38 please use this method by selecting Fax mode to be T 38 If the service provider does not support 38 pass through mode may be used To send or receive faxes in fax pass through mode users will need to select all the Preferred Codecs to be PCMU PCMA 5 5 LEDLight Pattern Indication Following tables show the LED light pattern indication RED LED always indicates not abnormal status Button flashes every 2 seconds 1f DHCP is configured Button flashes every 2 seconds if SIP server is configured Firmware Upgrading Button flashes every 2 seconds Device Malfunctions Red light steady on GREEN LED mostly indicates normal working status Message Waiting Indication Button flashes every 2 seconds 16 HandyTone 488 User Manual Grandstream Networks Inc RINGING Button flashes at 1 10 second RINGING INTERVAL Button flashes every second Green light steady on 17 HandyTone 488 User Manual Grandstream Networks Inc 6 Configuration Guide 6 1 Configuring HandyTone 488 WAN IP through Voice Prompt 6 1 1 DHCP Mode Follow section 5 1 with voice menu option 01 to enable HandyTone 488 to use DHCP 6 1 2 STATIC IP Mode Follow section 5 1 with voice menu option
52. uthenticate Account information password for ATA to register to SIP servers of ITSP Password 28 HandyTone 488 User Manual Grandstream Networks Inc Name Use DNS SRV User ID is Phone Number SIP Registration Unregister on Reboot Register Expiration Local SIP port Local RTP port Use Random Port Payload Type Send Send Flash Event Enable Call Features SIP service subscriber s name which will be used for Caller ID display Default is No If set to Yes the client will use DNS SRV to lookup for the SIP server If Yes is set a user phone parameter will be attached to the From header in SIP request This parameter controls whether the ATA needs to send REGISTER messages to the proxy server The default setting is Yes Default is No If set to yes the device will first send registration request to remove all previous bindings Use only if proxy supports this remove bindings request This parameter allows the user to specify the time frequency in minutes the HandyTone ATA refreshes its registration with the specified registrar The default interval 1s 60 minutes or 1 hour The maximum interval 1s 65535 minutes about 45 days This parameter defines the local SIP port the HandyTone ATA will listen and transmit The default value for FXS port 1s 5060 This parameter defines the local RTP RTCP port pair the HandyTone ATA will listen and tran

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