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User Manual - HipTel Broadband Phone

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1. gt SIP Settings eee VOIP SIP Server Address OP addiese or URL DEVICE STAT Outbound Prowy OP e URL ey A NE SIP User ID Assigned user ID or phone ramber peace i Car be sarne as or different from SIP User ID Authentication Password For secunty pess word does not display Hame Optional Use DNS SRV fio Yes User ID E phone number w e Ho c Yes SIP Registration s Yes No Unregister On Reboot Cys Ho Register Expiration lao Minutes Default is 1 hour max 45 daya Local SIP Port S060 Defot S060 Local RTP Port 5004 1024 65535 default 5004 Use Random Port No Yes C Ho HAT Treversal Yeo STUN server is URL or P port Keep Connected Interval o Seconde Default 20 seconde Use HAT IP af specified this IP address is used m SIPYSDP message fe ent eae ae 7 21 IPPH 202 User Manual 6 3 1 Determine your VoIP device s IP address On the IP PHONE202 while handset is up push the MENU key to obtain the phone s IP address which will show on the LCD screen If the IP address is all O s then your VoIP device does not have internet connection Please go back to step 1 and check your installation before proceeding 6 3 2 User Programming and Configuration From your web browser the IP PHONE 202 will show the following login screen Password voip or admin 6 3 3 Configu
2. version phone code 5 c 2004 05 06 date vocoder 6 1 0 0 6 version vocoder 7 h 2004 06 17 date web server 8 1 0 0 36 version web server 9 ir 2004 05 12 date 1st ring tone 10 1 0 0 0 version ring tone 11 2r 2004 05 12 date 2nd ring tone 12 1 0 0 0 version ring tone 13 3r 0000 00 00 date 3rd ring tone 14 0 0 0 0 version ring tone all show 0 means not usable or not supported Press Menu to exit Show 10 Phy Addr Press MENU to show MAC address MN Press or to exit Show 11 ring 0 Press MENU to hear current ring tone Press or to choose ring tones 4 ring tones available at present ringO default ring1 ring2 ring3 are not usable or not supported Press MENU to choose and save Be effective directly No need to restart Show rESEt ATTENTION A gt Enter MAC address and press MENU IP PHONE 202 will restart and get back to all default setting All former configuration will be lost B gt Press MENU and the phone will restart Others When in the mode e Press or and show ring 4 press or to adjust the ring volume form 0 to 7 e Press SPEAKER or pick up handset then press or to adjust voice volume 19 21
3. SPEAKER button 2 Press the REDIAL button to dial the Dial the last called number 3 View the CALLED history and choose to redial 4 View the CALLER history and choose to redial IP address dialing mode 1 Press MENU button to view the local IP address 2 Dial the IP address and press to start calling 5 2 2 Call Hold While in conversation pressing the Hold button will put the remote end on hold 4 21 IPPH 202 User Manual Pressing the Hold button again will release the previously Hold state and the bidirectional media will resume 5 2 3 Call Flash Assuming that call party A and party B are in conversation Party A wants to speak to party C or if there is another incoming call 1 Party A presses FLASH to get a dial tone 2 Party A then dials party C s number then Party A and party C now are in conversation 3 By pressing FLASH button Party A can choose to have conversation with B or C 5 2 4 Call Transfer 5 2 4 1 Blind Transfer The user can transfer an active call to a third party without announcement Assuming that call party A and party B are in conversation Party A wants to transfer party B to party Ci 1 Party A presses TRANSF to get a dial tone 2 Party A then dials party C s number do not press 3 Party A then press REDIAL A quick confirmation tone if the transfer has been successful Party B and party C now are in conversation 4 Party A can hang up or make
4. Subnet Mask T E E E Default Router T iT iT iT DMS Server 1 T fo fo fo DNS Server 2 lo iT P 7 Saveset Reboot Network Settings Dmamcaw Aes oned If DHCP mode is enabled then all the field values for the Static IP mode are not y y gned ised even though they are still saved in the Flash memory and the IP phone will i acquire its IP address from the first DHCP server it discovers on the LAN it 14 21 IPPH 202 User Manual attaches to PPPoE account settings can be configured here too if the user is using DSL ADSL connection Users can specify DNS server manually by entering DNS server s IP address If Static IP mode is selected then the IP address Subnet Mask Default Router IP address DNS Server 1 primary and mandatory DNS Server 2 Secondary and optional fields will need to be configured These fields are reset to zero by default Other settings fime Zie GKT DO US Ezstem Time New York GKT D US Eastem Time New Yok M SEER wipte EM PETRI Eh BASIE OPTION CTs Eset to Yes depky tone silbe L hoor head of noesa tine BASIC OPTION Cts ifi bo Yes depky tone ol be L hoor ahead af noesa tine E wa 3 iS K Teee fAneth Lesy tose iss amp Tex Hidhe aR fas MO Pas G Wie Pe ary Cheney Fama C Hidh Das Year A C tooth Derg Year sles Th p Ts s gt ce SUF SE E Tie 5 C tig ah Yar SUF SE Tle 5 C bag Mah Yar i Sesa a Set Rabe Other Settings This parameter
5. receiver mode to hand free mode press down the SPEAKER button while placing the receiver back to resting location Now the speakerphone is activated and you may continue speaking without receiver 5 4 Call Features Following table shows the new call features of IP PHONE 202 Not Disturb When someone call you do not disturb busy tone will be heard be use this feature pick up the handset and dial 70 It will be cancelled when you hang EE Call Forward To use this feature dial 72 and get the dial tone Then dial the forward number and for a dial tone and then hang up Cancel Unconditional Call Forward To cancel Unconditional Call Forward dial 73 and get the dial tone then hang up Busy Call Forward To use this feature dial 90 and get the dial tone Then dial the forward number and for a dial tone and then hang up Cancel Busy Call Forward To cancel Busy Call Forward dial 91 and get the dial tone then hang up Delayed Call Forward To use this feature dial 92 and get the dial tone Dial the forward number and for a dial tone and then hang up Cancel Delayed Call Forward To cancel this feature dial 93 get the dial tone and then hang up to a new channel to make a new call When in conversation this action will switch to the new incoming call if there is a call Flash Hook waiting indication When in conversati
6. ID Can be identical to or different from Account ID SIP User ID Authentication SIP service subscriber s account password for security the password does not Password display ame SIP service subscriber s name which will be used for Caller ID display Use DNS SRV Default is No If set to Yes then the client will use DNS SRV for server lookup If the IP PHONE 202 has an assigned PSTN telephone number then this field should be set to Yes Otherwise set it to No If Yes is set a user phone parameter will be attached to the From header in SIP request message to SIP server This parameter controls whether the IP PHONE 202 needs to send REGISTER messages to the proxy server The default setting is Yes wa en Default is No If set to Yes then the SIP user will be unregistered on reboot This parameter allows the user to specify the time frequency in minutes the IP PHONE 202 refreshes its registration with the specified gatekeeper SIP server The default interval is 60 minutes or 1 hour The maximum interval is 65535 minutes about 45 days This parameter defines the local SIP port the IP PHONE 202 listens and transmits Local SIP Port i signals The default value is 5060 This parameter defines the local RTP RTCP port pair the IP PHONE 202 will listen and transmit signals It is the base RTP port for channel 0 When Local RTP port configured channel 0 will use this port _value
7. MENU to show subnet address If DHCP is OFF enter new subnet address Press or to exit Press MENU to save or exit Restart to be effective Show 4 routEr Press MENU to show router gateway address If DHCP is OFF enter router gateway address Press or to exit Press MENU to save or exit Restart to be effective Show 5 dns Press MENU to show DNS address If DHCP is OFF enter new DNS address Press or to exit Press MENU to save or exit Restart to be effective Show 6 tFtP Press MENU to show tFtP address Enter new tFtP address Press or to exit Press MENU to save or exit Show 7 G 711u 2 Press MENU to show current codec Press or to view all codec options 1 G 711A 2 2 G 722 2 3 G 723 1 4 G 726 2 5 G 728 8 6 G 729 2 7 iLBC 1 18 21 IPPH 202 User Manual Press 1 to 9 to choose the frame number Press MENU to exit No need to restart Show 8 SIP SP 1 Press MENU to show SIP server service provider Press or to view all usable SIP server 1 9 Press MENU to save and exit SIP server must be configured through Web browser Show 9 codE rEL Press Menu to display the code releases Press or to view 1 b 2004 06 12 date boot code 2 1 0 0 18 version boot code 3 P 2004 06 21 date phone code 4 1 0 5 3
8. RARP ICMP DNS DHCP NTP TFTP protocols DHCP support for LAN or Cable modem PPPoE support for ADSL or Cable modem Supports IP address dynamic DHCP system Supports multi method NAT traversal and anti virus crossing Compliant with ITU T standard and DTMF form and check Supports codecs G 723 1 5 3K 6 3K G 729A B iLBC G 726 G 728 and wide band G 722 sound coding and decoding Supports dynamic sound check VAD Voice Activity Detection CNG Comfort Noise Generation Line Echo Cancellation G 168 and AGC Automatic Gain Control Dynamic sound technology with sound quality as good as traditional telephones Advanced Digital Signal Processing DSP to ensure superb hi fidelity audio quality Support standard voice features such as Caller ID Display or Block Call Waiting Hold Transfer Forward FLASH in band and out of band DTMF RFC2833 Dial Plans off hook auto dial configurable emergency dialing e g 911 early dial click to dial Advanced adaptive jitter buffer control packet delay amp loss concealment technology Support multi net construction Auto manual indication Built in world clock auto time display Has single or dual network ports 10 BaseT Support dial code regulation based on E 164 2 21 IPPH 202 User Manual Support TFTP remote upgrade English language IE setting Dialed received number check Speed dial function Emulate telephone dialing Easy volume adjustment IP to IP and point to point co
9. another call Both A and B can generate Blind Transfer 5 2 4 2 Attended Transfer The user can transfer an active call to a third party with announcement Assuming that call party A and party B are in conversation Party A wants to transfer party B to party C 1 Party A presses Flash to get a dial tone 2 Party A then dials party C s number then 3 If party C is answered press TRANSF to complete the transfer operation 4 If the call is not answered pressing flash button to resume the original call Both A and B can generate Attended Transfer 5 2 5 3 Way Conference Call Assuming that call party A called party B and now they are in conversation Party A wants to let party C to attend the conversation 1 Party A presses CONF to get a dial tone 2 Then press party C s number then or wait for 4 seconds Party A and party C now are in conversation 5 21 IPPH 202 User Manual 3 Party A presses CONF again now the 3 parties are all in conversation Both A and B can generate 3 way conference call Note The server needs to support this function for this feature or the phones to work 5 3 Receiving Calls When phone rings from incoming calls and to receive the incoming calls pick up the receiver or press the SPEAKER button After picking up the call to switch from hand free mode to receiver mode just pick up the receiver and begin talking the speakerphone will be deactivated To switch from
10. TTP server is listening to it can via HTTP server be omitted if using default port 80 Note Auto Upgrade has to be set to Yes to enable HTTP upgrade 13 21 IPPH 202 User Manual Choose Yes to enable HTTP upgrade and provisioning In Check for new Auto Upgrade firmware every ____ days field enter the number of days to check the HTTP server for firmware upgrade or configuration This parameter defines the URI or IP address of the NTP server which the IP phone will use to synchronize the current date time Default is No When set to Yes a SUBSCRIBE for Message Waiting SUBSCRIBE for MWI Indication will be sent periodically XS Impedance Selects the impedance of the analog telephone connected to the Phone port Lock Keypad If this parameter is set to Yes the configuration update via keypad is Update disabled Debug This parameter defines the URI or IP address of the Debug server Default is None Choose the debug level you want when using this function Default is voip or admin This contains the password to access the Advanced Web Configuration page This field is case sensitive Network Settings Yy I P BASIC OPTIONS gt Network Settings p Dynamically Assigned IF DHCP by default or PPPoE DEVICE STATUS PPPoE Account ID BASIC OPTIONS PPPoE Password OO Network Settings RRE Preferred DHS Server lo lo lo lo SUPER OPTIONS Static IP IP Address fi gq f 65 iT f 60
11. User Manual IPPH 202 User Manual IPPH 202 VoIP Telephone IPPH 202 User Manual Table of Contents le WELCOME wacestan ccd saanasiargennasngsnneseecausaaacazacaniuane tana mngsas gubegaah anaboas anes suansaiesacuumeanrennaianeinctaesaea 2 2 WHAT TS TIN TEE PACKAGE neinna ea teiuaine leds an OA areas 2 3 PRODUC F OVER VIEW iis eae T A AA 2 Dal KO POALULCS inr ecanseeicangasne T a E 2 5 2 HardWare specif ICAUIOM aniei a aa 2 As INSTALLATION cis atatssa te enaa E E O a 2 5 BASIC OPERATIONS iiyaa ain E T T A 2 5 Get Falnilial WIth FUNGCUOR KEYS seuncsaisocce ccnsasetuecueeeus con cra a bunctseaieaneniasegneanrebateestades 2 5 2 Dialing OF Making CaS neuint n onsacade yvetictanndatanens ven A aa Aa i 2 52 4 TWO dialing MOdES gt secessccsncvecucavene tere annone AE EON AANA 2 SPAT iE o E T E NE EP r T A A A E 2 IA CANFAS Na a e a T A 2 SZA Call ha SbCl enna yes a a a a E a E A 2 Pk EDING ans O aaa N E une icuetaeenecnesnass 2 5 2 4 2 Attended Manser mojari a a andes 2 Be RECEIVING Call E araa a a a a iesecedane eae euaaiceestad 2 5 4 Call REGUS ii A a T a 2 6 PROGRAMMING IP PHONE 202c rrireia nen anakina aaan A AEKA TN AeA NEKAT o AEAN 2 6 1 Default Administrator Password voip or admin 6 2 Major Sets of Configuration Data 2 6 3 Programming Using Web BrOoWSer sssanssnnnnnnnsnnnnnnnnsnennnnnnnnnnnnnenennnnnnnnnnnnnnnnnnnnenenennnnn 2 6 3 1 Determine your VoIP device s IP address ceccecseeee
12. controls how the displayed date time will be adjusted according to Time Zone the specified time zone This parameter controls whether the displayed time will be daylight savings time or not If set to Yes then the displayed time will be 1 hour ahead of normal time Date Display Format This parameter controls the date display format Bac es aod This contains the password to access the Basic Web Configuration page This field is case sensitive 15 21 IPPH 202 User Manual Device Status VOIP a MAC Address 00 09 45 70 88 86 WAN IP Address 192 168 1 126 DEVICE STATUS Software Version 2 0 6 5 2005 09 19 BASIC OPTIONS System Up Time lo dais 0 hone 2 marvatets SUPER OPTIONS a rs Registered Il Ho PPPoE Link Up disabled HAT detected HAT type is open Intemet HAT Mapped IP 00 06 NAT Mapped Port IE Total Inbound Calls l o Total Outbound Calls o Total Missed Calls Total Call Time 0 hvimmaates Total SIP Message Sent o Total SIP Message Received U Total RTP Packet Sent a Total RTP Packet Recerved o Total RTP Packet Loss ilo Reboot Device Status MAC Address The device ID in HEX format This is very important ID for ISP troubleshooting WAN IP Address This field shows WAN port IP address Software Version This field shows the software version System Up Time This shows system up time since last reboot x 6 3 4 Saving the Configuration Changes Once a change is made
13. e corresponding SIP address Enable Call Features aie is No If set to Yes Call Forwarding amp Do Not Disturb are supported 12 21 IPPH 202 User Manual Disable Call Waiting Default is No The user can setup up one number on each of the ring tones 1 2 and 3 Whena call is received from one of these numbers the respective ring tone will be generated Other Settings Yig f ioo PIKI oY i Seime it i a 7 Other Settings The IP of TFTP server needs to be configured If it is non zero or not blank the IP phone will attempt to retrieve new configuration file or new code image from the specified tftp server at boot time It will make up to 3 attempts before timeout and then it 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 will be verified and then saved into the Flash memory Be careful when the TFTP is in progress the phone back light on LCD screen will be up Do NOT interrupt the process especially the power supply otherwise might cause the device dead Be patient as in some network this process will take up to 15 minutes Firmware Upgrade via TFTP server The URL for the HTTP server used for firmware upgrade and configuration via HTTP For example http upgrade myVolIPservice com 6688 IP PHONE202 Firmware Upgrade Here 6688 is the specific TCP port that the H
14. eeeteeseeseeeeeeeeeesseseteeseeseenaeeetnes 2 6 3 2 User Programming and Configuration cccccccsscseeeeeeeseeseesaeeeeeesaeeaeeaeeaeeaeeaeegeegs 2 6 3 3 Configuration Options and Explanations cccccscsecseeeeeeeseeeeeeeeeeeeesesseeaeeetaeseesaeeass 2 6 3 4 Saving the Configuration Changes ccccccceeceeeeseeeeeeeeseeeeesaesaeeaeeaesaesaeeaeeaeeaeeaeees 2 6 3 5 Rebooting the IP PHONE 202r sdara aller ate teed e a fh aaa 2 6 4 Programming Using Keypad s ssssssssnnsnsrnnnsnnnnnnnnnrnrnnnnnnrnnnnnnrnrnnnnrnrnnennnnnnenenrnnnnennnnnnnnn 2 Te TEST YOUR SETUP ciiai 2 Oe VV AR RAINEY a A N Gun uenannaianats 2 1 21 IPPH 202 User Manual 1 WELCOME Prescient Worldwides s IP PHONE 202 is an internet telephone set that features superb audio quality rich functionalities high level of integration and compactness By converting analog voice for transmission over the internet the IP Phone 202 allows users with broadband internet connections to make calls to and from anywhere in the world The IP PHONE 202 is fully compatible with SIP industry standard and can interoperate with many other SIP compliant devices and software on the market 2 WHAT IS IN THE PACKAGE The IP PHONE 202 package contains 1 One IP PHONE 202 VoIP phone 2 One IP PHONE 202 phone handset 3 One universal power supply 4 One Ethernet cable 3 PRODUCT OVERVIEW 3 1 Key Features Supports SIP 2 0 Supports TCP UDP IP RTP RTCP HTTP ARP
15. es Settings arly Dial Default is No Use only if proxy supports 484 response ial Plan Prefix Sets the prefix added to each dialed number No Key Entry Default is 4 seconds This is the time lapse after entering a phone number for Timeout the IP PHONE 202 to begin completing your call This parameter allows users to configure the key to be used as the Send Enter or Dial key If set to Yes pressing this key will immediately trigger the sending of dialed numbers entered thus far In this case this key is essentially equivalent to the Re Dial key If set to No this key will then be included as part of the dial string to be sent out Use as Dial Key Auto Answer Voice Default is NO When set to yes the phone will automatically pick up Mail the call and turn the speaker on This parameter defines the User ID or extension number of a 3rd party voice mail system where the user may have an account By Voice Mail UserID defining this Voice Mail extension when the user presses the Message button on the phone an INVITE message will be sent to that extension to allow the user to retrieve messages This parameter allows users to configure a User ID or extension number to be automatically dialed upon off hook Please note that only the user part of a SIP Off hook Auto Dial jaddress needs to be entered here The IP PHONE 202 will automatically append the and the host portion of th
16. for RTP and the port_value 1 for its RTCP channel 1 will use port_value 2 for RTP and port_value 3 for its RTCP The default value is 5004 SIP Registration Register Expiration When set to Yes this parameter will force random generation of both the local SIP and RTP ports This is usually necessary when multiple IP PHONE 202 are behind the same NAT Use Random Port detected NAT is a Full Cone Restricted Cone or a Port Restricted Cone the IP PHONE 202 will attempt to 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 This parameter defines whether the IP PHONE 202 NAT traversal mechanism will be activated or not If activated by choosing Yes and a STUN server address is also specified then the IP PHONE 202 will automatically follow the STUN client specifications Under this mode the embedded STUN client inside NAT Traversal the IP PHONE 202 will attempt to detect if there is a and type of firewall NAT it is sitting behind through communication with the specified STUN server If the 9 21 IPPH 202 User Manual specified STUN server the IP PHONE 202 will periodically every 20 seconds send a blank UDP packet with no payload data to the SIP server to keep the hole on the NAT open This parameter specifies how often the IP PHONE 202 sends a blank UDP packet to the SIP server in order to keep the hole on
17. mmunication HiFi low delay net sound communication Local IP number local number check Instant plug and play Full duplex hands free speakerphone superior frequency Call forwarding call waiting call hold call transfer redial call log volume control voice mail with indicator 3 2 Hardware Specification The table below lists the hardware specification of IP PHONE 202 PC interface Available for IP PHONE 202B Universal Switching Power Supply Input 100 240VAC 50 60 Hz Output 9VDC 600mA WAN interface 1 x RJ45 10 Base T roduct Color Dimension All Package Weight 1 01kg Operation Temperature 0 40 C 32 104 F torage Temperature 30 65 C 22 149 Humidity pos 4 INSTALLATION 1 Insert power adapter into back of the phone and connect it to a power outlet 2 Connect internet cable from back of the phone WAN Port to broadband modem router hub or switch 3 For IP PHONE 202B connect internet cable from back of the phone PC Port to a PC 3 21 IPPH 202 User Manual Warning Please do not use a different power adapter Using other power adapter may damage the IP PHONE 202 and will void the manufacturer warranty 5 BASIC OPERATIONS 5 1 Get Familiar with Function Keys volume of the loudhailer a een volume of the loudhailer Enter the hand free mode 5 2 Dialing or Making Calls 5 2 1 Two dialing modes Ordinary dialing mode 4 ways 1 Pick the up handset or press the
18. nes the size of the iLBC codec frame The default setting is 20ms This defines payload time for iLBC Default value is 97 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 codec in the above codec Preference List or the actual used payload type negotiated between the 2 conversation parties at run time For example if the first codec 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 is set to be 2 and if the first codec chosen is G729 or G711 or G726 then the ptime 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 IP PHONE 202 will use and save the maximum allowed value for the corresponding first codec choice The maximum value for PCM is 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 Layer 3 QoS This field defines the layer 3 QoS parameter which can be the value used for IP y Precedence or Diff Serv or MPLS Default value is 48 Layer 2 QoS This co
19. ntains the value used for layer 2 VLAN tag Default setting is 0 This controls the silence suppression VAD 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 to make the connection feel more natural If set to No this feature is disabled Silence Suppression 11 21 IPPH 202 User Manual Phone Features Settings VOIP SUPER OPTIONS Phone Features Settings Early Dial No Yes Select Yes only if proxy supports 424 response DEVICE STATUS Dial Plan Prefix This prefix string is added to each dialed number No Key Entry Timeout a seconds Default is 4 seconds Use a Dial Rey C Ho Yes lf zet to Yes will function as the Redial hey SIF Settings Auto Answer No C Ye Audio Sethngs Phone Features Other Settings Voice Mail UserID ae nea KUser Diextension for 3rd party voice mail system Qithook Avato Dial User Diextension to dial automatically when offhook Enable Call Features ds to stn LE Yes Call Forwarding amp Call Waiting Disable are supported locally Disable Call Waiting No CYs sveterm ring tone custom ring tone 1 used if incoming caller ID is Default Ring Tore i ji C custom ring tons 2 used if incoming caller ID is custom ring tone 3 used if incoming caller ID is Reboot Phone Featur
20. on without an incoming call this action will switch 6 PROGRAMMING IP PHONE 202 6 21 IPPH 202 User Manual 6 1 Default Administrator Password voip or admin 6 2 Major Sets of Configuration Data There are 8 major sets of data related to the programming of the IP Phone 202 IP Type IP Address Subnet Mask Router DNS Server1 DNS Server2 Account and Password The IP Phone 202 needs to be assigned an IP address in order for it to access the internet and make telephone calls There are 2 kinds of IP address types IP Type 1 IP Type O Static IP address is manually entered by you There are 5 fields to be filled IP Address 192 168 0 160 default Subnet Mask 0 0 0 0 Default Router 0 0 0 0 DNS Server1 DNS using 0 0 0 0 DNS Server2 DNS spare 0 0 0 0 2 IP Type 1 DHCP IP address is dynamically assigned by router if any at the location thereby allowing it to access the internet and make calls Plug the IP Phone 202 into the router using the LAN port connection The IP address is assigned to the IP Phone 202 automatically 3 User Account Assign your own user name 4 User Password Assign your own password 6 3 Programming Using Web Browser IP PHONE 202 has embedded Web server and HTML pages that allow users to configure the IP PHONE 202 through an easy to use Web browser interface such as Microsoft s Internet Explorer or Netscape browser Below is a screen shot of the IP PHONE 202 configuration page SUPER OPTIONS
21. ration Options and Explanations After a correct password is entered in the login screen the embedded web server inside the IP PHONE 202 will show the configuration page which is explained in details below SIP Settings VOIP BASIC OPTIONS SUPER OPTIONS gt SIP Settings SIP Server Address Outbound Prosar SIP User ID IP address or URL IP address or URL af any Assemed user ID or phone ruimber Account ID Avtthentication Password Car be same as or different from SIP User ID Hame Use DNS SRY User ID amp phone number SIP Registration Unregister On Reboot Register Expiration e0 Minutes Default is 1 hour max 45 daya For secunty pessword does not display L Optional No C Yes Yee C No E Mo No Yeg C Yes Local SIP Part Local RTF Port Use Random Fort NAT Traversal Keep Connected Interval Use HAT IF S060 Default 5060 5004 1024 65535 default 5004 Ho Yes No i Yes STUH server zl URL or IP yport o Seconds Defawli 20 seconds f specifiad this IF address is wed m SIPSDE meseage 8 21 IPPH 202 User Manual SUPER USER OPTIONS PAGE IP Server Address Gatekeeper s SIP Server s URL or IP address utbound Proxy SIP Outbound Proxy Server s URL or IP address IP User ID SIP service subscriber s User ID SIP service subscriber s Authenticate
22. the NAT open NAT IP address used in SIP SDP message Default is blank SIP Extension to notify SIP server that the unit is behind the NAT Firewall This parameter controls how DTMF events are transmitted There are 3 ways In Audio which means DTMF is combined in audio signal not very reliable with low bit rate codec via RTP RFC2833 or via SIP INFO This parameter sets the payload type for DTMF using RFC2833 If this parameter is set to Yes the From header in outgoing INVITE message will be set to anonymous essentially blocking the Caller ID from displaying This parameter allows users to control whether to send a SIP NOTIFY Send Flash Event message indicating that the phone s Flash key has been pushed or just to switch to the voice channel when users press the Flash key Audio Settings Ty j p Bu PRR OP Wet ASAT T sere PRB IGE STALL 10 21 IPPH 202 User Manual The IP PHONE 202 supports up to 7 different Codec types including G 711 G 723 1 G 726 G 728 G 729A B iLBC Users can configure Codecs in a preference list that will be included with the same preference order in SDP message The first Codec in this list can be entered by choosing the appropriate option in Choice 1 Similarly the last Codec in this list can be entered by choosing the appropriate option in Choice 7 Preferred Codec This defines the encoding rate for G723 codec By default 6 3kbps rate is chosen a This defi
23. users should click on the SaveSet button in the Configuration page 16 21 IPPH 202 User Manual The IP PHONE 202 will then display the following screen to confirm that the changes have been saved Please wait 5 seconds and then reboot the device DEVICE STATUS DEVICE STATUS BASIC CPTIONS SUPER OFTIONS Your changes have heen saved Toor changes have been saved Please wat 3 second and then ceboct the dence Please wat 3 second and then ceboct the dence 6 3 5 Rebooting the IP PHONE 202 You can reboot the IP PHONE 202 by clicking on the Reboot button after each update to the configuration page Alternatively you can reboot by unplugging the power supply of the IP PHONE 202 and then powering it on again If your IP PHONE 202 ever becomes stuck or un responsive you can unplug the power supply to reboot it Frequent rebooting by unplugging the power supply is not recommended and should not be necessary 17 21 IPPH 202 User Manual 6 4 Programming Using Keypad Configure with keypad a Show 1 dhcp on Press 1 dhcp off as the current option Press MENU and enter the menu mode Press or to choose Press MENU to save or exit Restart to be effective Show 2 IP Addr Press MENU to show the current IP address If DHCP is OFF enter new IP address Press or to exit Press MENU to save or exit Restart to be effective Show 3 SubNet Press

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