VoIPIngate User Manual
Contents
1. Via HTTP Server The URL for the HTTP server used for firmware upgrade and configuration via HTTP For example http provisioning mvcompanv com 6688 Grandstream 1 0 5 16 Here 6688 is the specific TCP port that the HTTP server is listening at it can be omitted if using default port 80 Note If Auto Upgrade is set to No VoIP Client ATA will only do HTTP download once at boot up Automatic HTTP Choose Yes to enable automatic HTTP upgrade and provisioning Upgrade In Check for new firmware every field Enter the number of days period VoIP Client ATA will check the HTTP server for firmware upgrade or configuration after the defined number of days When set to No VoIP Client ATA will only do HTTP upgrade once at boot up SUBSCRIBE for Default is No When set to Yes a SUBSCRIBE for Message Waiting MWI Indication will be sent periodicallv Offhook This parameter allows the user to configure a User ID or extension number to Auto Dial be automatically dialed upon offhook Please note that only the user part of a SIP address needs to be entered here The phone will automatically append the and the host portion of the corresponding SIP address Enable Call Feature Default is No If set to Yes Call Forwarding amp Do Not Disturb are supported locally Disable Call Default is No Waiting Send DTMF This parameter controls the way DTMF events are transmitted There are 3 ways in2. key will then be included as part of the dial string to be sent out Local SIP port This parameter defines the local SIP port the IP phone will listen and transmit on The default value is 5060 This parameter defines the local RTP RTCP port pair the IP phone will listen Local RTP port and transmit on 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_value 2 for RTP and port_value 3 for its RTCP The default value is 5004 ee Randani Gr This parameter when set to Yes will force random generation of both the local SIP and RTP ports This is usually necessary when multiple IP phones are behind the same NAT keep alive interval The VoIP Client ATA sends a UDP package to the SIP server periodically in order to keep the port open on the router This parameter defines the interval time that HT286 send the UDP package The default setting is 20 second Use NAT IP NAT IP address used in SIP SDP message Default is blank Proxy Require SIP Extension to notify SIP server that the unit is behind the NAT Firewall NAT Traversal This parameter defines whether the phone NAT traversal mechanism will be activated or not If activated by choosing Yes and a STUN server is also specified then the phone will behave according to the STUN client specification Under this mode the embedded STUN client inside the phone wi
3. 2005 VoIP Client ATA 8 02 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 19 Cancel Delaved Call Forward To cancel this Forward dial 93 and get the dial tone then hang up Flash Hook When in conversation this action will switch to the new incoming call if there is a call waiting indication When in conversation without an incoming call this action will switch to a new channel for a new call 2 4 Fax Support VoIP Client ATA supports FAX in two modes T 38 Fax over IP and fax pass through T 38 is the preferred method because it is more reliable and works well in most network conditions If the service provider supports T 38 please use this method by selecting Fax mode to be T 38 If the service provider does not support T 38 pass through mode may be used To send or receive faxes in fax pass through mode users will need to select all the Preferred Codecs to be PCMU PCMA 2 5 LED Light Pattern Indication Following are the LED light pattern indications RED LED indicates abnormal status DHCP Failed or WAN No Cable flash every 2 seconds if DHCP is configured VOIP Client 486 fails to register flash every 2 seconds if SIP is configured GREEN LED indicates normal working status Message Waiting Indication Button flashes every 2 seconds RINGING Button flashes at 1
4. PHONE CaS cs tas tii ace a Amin i A 6 2 2 1 Calling phone or extension numbers 2 2 2 Direct IP calls 2 23 Blind Wansfet iii i B sapassie ees ysibeerinses pele es 7 22 4 Attended trans ter er si ia L A EE AARAA BIE e ina az 8 2 32 Call Transfers irina eer sa wa b BSE TRABI SA SE PER KAS SA 8 2 RAK SUPPO ces Naat a e g L Dave b EEA EO AE a a A 9 2 5 Led Light Pattern Indications ii xiiutsi se a nerd eadtsess sea 9 3 WONT uraaoii G de ia a Ki i E ET aa 10 3 2 Configuring VOIP Client with Web Browser 3 2 1 Access the Web Configuration Menu 3 22 End Wer Comipuranon sis i ee ees eed Pee 11 3 2 3 Advanced User Configuration 22s seen nterna 14 3 2 4 Saving the Configuration Changes LLs e 24 3 2 5 Rebooting the VoIP Client ATA from Remote 22 mn 24 4 Restoring Factory Default Settings ius ea pa A PN 23 Voip Ingate Tithe Page sc ses ccs sevnsescvsdecdcpiainvcneade in kebbes pg 26 Introduction What it is and How it Works sess eseennenennzznnenzznnenzznnnnzzzznnznnanazza 2l Chapter 1 Set up and Installation ssesseesesesseseeeseeeeeseeseseresetsstserssresstssrssressersresreesesee 28 Chapter 2 Installing the Manager sgassat ev encvercnesouncees donleteteavtentaness 30 Install the Voip Ingate Manager Application ss nsnsennzennsenzzznnznzenmazznnnaaza 30 Define the Comport Connectikti fire e 33 Chapter 3 Port and SIM Settings Jciissavesassisnecaisasiavsaeasa geaetaseavassiseseaecsuesaecossnczasdee
5. SSeS PPPoE password Preferred DNS server e Je fe e E statically configured as IP Address we tes a fo Subnet Mask 255 255 255s o Default Router pefe ja fi DNS Server 1 2 jjs fi DNS Server 2 BKIK May 30 2005 VoIP Client ATA 11 current setting is GMT 5 00 US Eastern Time New York Daylight Savings E yy UB yes if set to Yes display time will be 1 hour ahead of normal Time time End User Password This contains the password to access the Web Configuration Menu This field is case sensitive with max 25 characters IP Address There are 2 modes under which the IP phone can operate If DHCP mode is enabled then all the field values for the Static IP mode are not used even though they are still saved in the Flash memory and the IP phone will acquire its IP address from the first DHCP server it discovers on the LAN it attaches to To use PPPoE feature please set the PPPoE account settings if the HT 286 is connected directly to a DSL modem The HT 286 will attempt to establish a PPPoE session if any of the PPPoE fields is set In this mode the WAN side web access is disabled and TFTP upgrade for firmware is not feasible and HTTP upgrade is the only available solution If Static IP mode is selected then the IP address Subnet Mask Default Router IP address DNS Server 1 primary DNS Server 2 secondary fields will need to be configured These fields are reset to zer
6. above message 3 2 5 Rebooting the VoIP Client ATA from a Remote Location The administrator of the phone can remotely reboot the phone by pressing the Reboot button at the bottom of the configuration menu Once done the following screen will be displayed to indicate that rebooting is underway The device is rebooting now You may relogin by clicking on the link below in 30 seconds Click to relogin At this point the user can relogin to the phone after waiting for about 30 seconds May 30 2005 VoIP Client ATA 24 4 Restoring Factory Default Settings Warning Restoring the Factory Default Settings will DELETE all configuration information of the device Please backup or print out all the settings before attempting the following steps Please disconnect the network cable and power cycle the unit before trying to reset the unit to factory default The steps are as follows e Step 1 Find the MAC Address of the device The MAC address of the device is located at the bottom of the device It is a 12 digits hex number e Step 2 Encode the MAC address to decimal digits Please use the following mapping 0 9 0 9 A 22 B 222 Cz 2222 D 33 E 333 F 3333 For example for MAC address 000b8200e395 the user encoding should be 0002228200333395 e Step 3 Access the voice menu by pressing or the LED button then dial 99 and get the voice prompt RESET e Step 4 Key in the enco
7. security protection SIP Server ESES e g sip mycompany com or IP address Outbound Proxy e g proxy myprovider com or IP address if any SIP User ID bra the user part of an SIP address Authenticate ID 2125250 can be identical to or different from SIP User ID Authenticate hii 4 M Password purposely not displayed for security protection Name optional e g John Doe Advanced Options Preferred Vocoder k current setting is PCMU in listed order B l current setting is PCMA choice 2 current setting is G723 choice 3 B current setting is G729 choice 4 fi current setting is G726 32 choice 5 2 current setting is G728 choice 6 5 2 current setting is iLBC choice 7 ku 6723 rate E 6 3kbps encoding rate u 5 3kbps encoding rate iLBC frame size UU 0m E 30ms iLBC payload type ee 96 and 127 default is 97 Silence Suppression E No G Yes Voice Frames pd a up to 10 20 32 64 for G711 G726 G723 other codecs respectively Fax Mode Ei 38 autoDetect E Pass Through Layer 3 QoS a Diff Serv or Precedence value May 30 2005 VoIP Client ATA 15 Layer 2 QoS 802 1Q VLAN Tag l 802 1p priority valuel 0 7 Use DNS SRV BB yo E Yes User ID is phone p number Ms No Yes SIP Registration Unregister On Reboot Register Expiration in minutes default 1 hour max 45 days Early Dial BB No U yes
8. 10 second RINGING INTERVAL Button flashes every second May 30 2005 VoIP Client ATA 9 3 Configuration Guide 3 2 Configuring VOIP Client with a Web Browser VoIP Client ATA has an embedded Web server that will respond to HTTP GET POST requests It also has embedded HTML pages that allow a user to configure the IP phone through a Web browser such as Microsoft s IE and AOL s Netscape 3 2 1 Access the Web Configuration Menu First get the IP address of the VOIP Client through section 2 1 with menu option 02 Then access the VOIP Client s Web Configuration Menu using the following URI http Phone IP Address where the Phone IP Address is the IP address of the phone May 30 2005 VoIP Client ATA 10 3 2 2 End User Configuration Once this request is entered and sent from a Web browser the IP phone will respond with the following login screen Password E The password is case sensitive with a maximum length of 25 characters The factory default password for End User is 123 or blank After the correct password is entered in the login screen the embedded Web server inside the IP phone will respond with the following Basic Settings configuration page which is explained in details below End U max b P ima purposely not displayed for security protection IP Address G dvnamicallv via DHCP default or PPPoE will attempt PPPoE if DHCP fails and following is non blank PPPoE account ID
9. RADIO TERMINAL VolPIngate a VoIP GSM gateway Connecting Cellular Phones Directly to Voice over Internet Protocol User Manual For more assistance contact www radioterminal ru Version 2 May 30 2005 1 High voltage transients surges and other power irregularities can cause extensive damage It is the user s responsibility to provide a power protection system WARNINGS 2 It is the user s responsibility to install operate and maintain a the system in accordance with all applicable codes regulations and safety measures Without prior notice and without obligation the contents of this manual may be revised to incorporate changes and improvements Every effort has been made to ensure that the information is complete and accurate at the time of publication Nevertheless Discovery Communication cannot be held responsible for errors or omissions TRADEMARKS AND PATENTS Trademarks patents and copyrights apply May 30 2005 Master VoIP i Table of Contents Table of Contents arc o a A A lhe Be SE lii Contents of io 6 0 EA EJT 1 VoIP Client ATA Title Page and Message ss sess essoseeseznnnznznnenzzznnnanzznntii 2 l Product OVErVie W eei soe os sce A ade eas 3 1 1 Key Features 1 2 Hardware Specifications 2 Basic Operations 2 1 Get familiar with keypad and voice prompt e eee eee eset teen eee ee en 5 2 2 Placing
10. aendeaaes 35 Dial Settings for tlie GSM Botti iri jie e g A e eae de 35 SM A MI EA A 36 Follow Me Setings i205 era ba geet A A ah a data ka 37 Call Back Setting Sni teneranu a e e e aare arial 38 May 30 2005 Master VoIP iii Contents of this Package The VoIP master package contains the following components e 1 Voip Ingate gateway e 1 antenna e 1 Voip Ingate software CD e 1 100 220V 24V transformer e 1 Com Port cable e 1 electric power cable e This manual is located on our Web site at www radioterminal in the Radioterminal section Check this site for updates to this manual May 30 2005 Voip Ingate VoIP Client ATA Before using this device please read the following 1 Connnect the VoiPMaster to the network You must have an account with a VoiP Provider or you should register an extension with a SIP Gateway Server Get all needed data from your provider such as user name Password server IP address ports etc 2 Connect a regular analog telephone to the system and configure it first as a regular VoiP client That configuration is done using a web interface You will find instructions on page 14 of this manual 3 Make sure you can make and receive calls using your regular phone set 4 Run the GSM management software and configure it according to the manual and the interface menu May 30 2005 VoIP Client ATA 2 1 1 1 Product Overview Key Features Suppor
11. ain software release This number is always used for firmware upgrade Bootloader This is normally not changed HTML This is the user interface normally not changed VOC This is the codec program normally not changed DIED UTES This shows system up time since last reboot Pecistered This shows whether the unit is registered to service provider s server EERDE HOK Up This shows whether the PPPoE is up if connected to DSL modem NAT This shows what kind NAT the VoIP Client ATA is connected to via its WAN port It is based on STUN protocol DA Doppa ik WAN side public IP if connected to LAN of a SOHO router NAT Mapped tor External port detected by STUN Statistical Status Self explainable Please refer to the page displayed 3 2 3 Advanced User Configuration To login to the Advanced User Configuration page follow the instruction in section 3 2 1 they will lead You to the following page The password is case sensitive with a maximum length of 25 characters and the factory default password for Advanced User is admin Password May 30 2005 VoIP Client ATA 14 Advanced User configuration page includes not only the end user configuration but also some advanced settings such as SIP configuration Codec selection NAT Traversal Setting and other miscellaneous settings Following is the screen shot of the Advanced configuration page Admin Password purposely not displayed for
12. audio which means DTMF is combined with the audio signal not very reliable with low bit rate codec via RTP RFC2833 or via SIP INFO DTMF Payload This parameter sets the payload type for DTMF using RFC2833 Type May 30 2005 VoIP Client ATA 21 Send Flash Event This parameter allows the user to control whether to send an SIP NOTIFY message indicating the Flash event or just to switch to the voice channel when the user presses the Flash key 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 e Bellcore North America e ETSI FSK France Germany Norway Taiwan UK CCA e ETSI DTMF Finland Sweden e DIMF Denmark Onhook Voltage Select the onhook voltage to suit different area or PBX Polaritv Reversal Select Polarity Reversal to adapt some call charge billing system Default is No NTP server This parameter defines the URI or IP address of the NTP server which the IP phone will use to display the current date time Send Anonymous If this parameter is set to Yes the From header in the outgoing INVITE message will be set to anonymous essentially blocking the Caller ID from being displayed Lock keypad If this parameter is set to Yes the configuration update via keypad is update disabled Route mae The IP address
13. ce current setting is 600 Ohm North America Caller ID Scheme current setings Becr Onhook Voltage current setings 36v Polarity Reversal Ea yo E Yes reverse polarity upon call establishment and termination NIP Server time nist gov URI or IP address Send Anonymous EH No UB yes caller ID will be blocked if set to Yes Lock keypad update yy Li Yes configuration update via keypad is disabled if set to Yes Syslog Server TENA Syslog Level current setting is INFO sl Administrator password Only the administrator can configure the Advanced Admin Password Settings page Password field is purposely left blank for security reasons after clicking update and saved The maximum password length is 25 characters This field contains the URI string or the IP address and port if different from 5060 of the SIP proxy server e g the following are some valid examples sip my voip provider com or sip my company sip server com or 192 168 1 200 5066 SIP Server This field contains the URI string or the IP address and port if different from Outbound Proxy 5060 of the outbound proxy If there is no outbound proxy this field SHOULD be left blank If not blank all outgoing requests will be sent to this outbound proxy May 30 2005 VoIP Client ATA 17 SIP User ID This field contains the user part of the SIP address for this phone e g if the SIP address is sip my_user_
14. ded MAC address decimal digits after hearing the IVR prompt Once the correct encoded MAC address is entered the device will reboot automatically and restore the factory default settings May 30 2005 VoIP Client ATA 25
15. ered by choosing the appropriate option in Choice 7 Silence Suppression This controls the silence suppression VAD feature of G723 and G729 If set to Yes when a silence is detected a small quantity of VAD packets instead of audio packets will be sent during the period of no talking If set to No this feature is disabled Layer 3 QoS This field defines the layer 3 QoS parameter which can be the value used for IP Precedence or Diff Serv Default value is 48 Layer 2 QoS This setting includes two fields The 802 1Q VLAN Tag contains the value used for layer 2 VLAN tag Default setting is blank And 802 1p priority value contains the value of the priority value Use DNS SRV This parameter controls whether the IP phone supports the DNS SRV route function May 30 2005 VoIP Client ATA 18 Voice Frames per This field contains the number of voice frames to be transmitted in a single TX 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 beca
16. es To make a direct IP call first pick up the analog phone or turn on the speakerphone on the analog phone then access the voice menu prompt by dial or press the button on the HT286 and dial 4T to access the direct IP call menu User will hear a voice prompt Direct IP Calling and a dial tone Enter a 12 digit target IP address to make a call The follow is a table of the encoding scheme for the most commonly used characters May 30 2005 VoIP Client ATA 6 INPUT Encoding 00 0l 02 03 04 05 06 07 08 O OO NI DN NM AI InN oO 09 0 dot character 4 column character Examples If the target IP address is 192 168 0 160 the dialing convention is Voice Prompt with option 47 then 192168000160 followed by pressing the key if it is configured as a send key or wait 4 seconds In this case the default destination port 5060 is used if no port is specified If the target IP address port is 192 168 1 20 5062 then the dialing convention would be Voice Prompt with option 47 then 192168001020 45062 followed by pressing the key if it is configured as a send key or wait for 4 seconds 2 2 3 Blind Transfer Assuming that call party A and B are in conversation A wants to Blind Transfer B to C 1 A presses FLASH on the analog phone or Hook Flash for old model phones to get a dial tone 2 Then A dials 87
17. ial tone 2 A then dial C s number then or wait for 4 seconds A and C now are in conversation 3 A can hang up Note 2 3 When intended Transfer failed if A hangs up the HandTone 496 will ring user A again to remind A that B is still on the call by pressing FLASH or Hook again will restore the conversation between A and B Call Features Following table shows the call features of VoIP Client ATA Key Call Features 30 Block Caller ID for all subsequent calls 31 Send Caller ID for all subsequent calls 67 Block Caller ID per call 82 Send Caller ID per call 50 Disable Call Waiting for all subsequent calls 51 Enable Call Waiting for all subsequent calls 70 Disable Call Waiting Per Call 71 Enable Call Waiting Per Call Unconditional Call Forward 72 To use this feature dial 72 and get the dial tone Then dial the forward number and for a dial tone then hang up Cancel Unconditional Call Forward 73 To cancel Unconditional Call Forward dial 73 and get the dial tone then hang up Busy Call Forward 90 To use this feature dial 90 and get the dial tone Then dial the forward number and for a dial tone then hang up 0 Cancel Busy Call Forward To cancel Busy Call Forward dial 91 and get the dial tone then hang up May 30
18. id my_provider com then the SIP User ID is my_user_id Please do NOT include the preceding sip scheme or the host portion of the SIP address in this field SIP User ID User account information provided by VoIP service provider ITSP usually has the digit form of a phone number or is actually a phone number Authenticate ID SIP service subscriber s ID used for authentication Can be identical to or different from SIP User ID Authenticate SIP service subscriber s account password for GXP 2000 to register to SIP Password servers of ITSP Name SIP service subscriber s name which will be used for Caller ID display G723 Rate This defines the encoding rate for G723 vocoder By default 6 3kbps rate is chosen iLBC frame size This defines the size of the iLBC codec frame The default setting is 20ms iLBC payload type This defines the iLBC payload type The default setting is 97 Preferred Vocoder VoIP Client ATA supports up to 7 different vocoder types including G711 ulaw PCMU G71 1 alaw PCMA G723 G729A B G726 32 ADPCM G728 and iLBC Depending on the product model some of these vocoders may not be provided in a standard release A user can configure vocoders in a preference list that will be included with the same preference order in SDP message The first vocoder in this list can be entered by choosing the appropriate option in Choice 1 Similarly the last vocoder in this list can be ent
19. ll attempt to detect if and what type of firewall NAT it is behind through communication with the specified STUN server If the detected NAT is a Full Cone Restricted Cone or a Port Restricted Cone the phone will attempt to use its mapped public IP address and port in all the SIP and SDP messages it sends out If this field is set to Yes with no specified STUN server then the phone will periodically every 20 seconds by default send a blank UDP packet with no payload data to the SIP server to keep the hole on the NAT open Firmware Upgrade This radio button will enable VoIP Client ATA to download firmware or configuration file through either TFTP or HTTP May 30 2005 VoIP Client ATA 20 Via TFTP Server This is the IP address of the configured tftp server If it is non zero or not blank the IP phone will attempt to retrieve new configuration file or new code image update 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 Note DO NOT interrupt the TFTP upgrade process especially the power supply as this will damage the device Depending on the network environment this process can take up to 15 or 20 minutes
20. o by default Time Zone This parameter controls how date time will be displayed according to the specified time zone Daylight Savings Time 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 In addition to the Basic Settings configuration page the end user also has access to the device Status page The following is a screen shot of the device Status page May 30 2005 VoIP Client ATA 12 Here are the details MAC Address 00 0B 82 01 56 4D WAN IP Address 192 168 1 12 Product Model HT286 FAN eae ee rae 1 0 6 3 Bootloader 1 0 1 0 HTML 1 0048 VOC System Up Time 0 day s 0 hour s 4 minute s Registered Yes PPPoE Link Up disabled NAT detected NAT type is full cone NAT Mapped IP 24 12 198 35 NAT Mapped Port 54060 Total Inbound Calls Total Outbound Calls Total Missed Calls Total Call Time in minutes Total SIP Message Received Total RTP Packet Sent Total RTP Packet Received 0 0 0 0 Total SIP Message Sent 5 5 0 0 Total RTP Packet Loss 0 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 Product Model This field contains the product model info May 30 2005 VoIP Client ATA 13 Software Version Program This is the m
21. or URL of the System log server This feature is especially useful ystos for ITSP Internet Telephone Service Provider May 30 2005 VoIP Client ATA 22 Syslog Level Select the ATA to report the log level Default is NONE The level is one of DEBUG INFO WARNING or ERROR Syslog messages are sent based on the following events e product model version on boot up INFO level e NAT related info INFO level e sent or received SIP message DEBUG level e SIP message summary INFO level e inbound and outbound calls INFO level e registration status change INFO level e negotiated codec INFO level e Ethernet link up INFO level e SLIC chip exception WARNING and ERROR levels e memory exception ERROR level The Syslog uses USER facility In addition to standard Syslog payload it contains the following components GS LOG device MAC address error code error message Here is an example May 19 02 40 38 192 168 1 14 GS_LOG 00 0b 82 00 a1 be 000 Ethernet link is up May 30 2005 VoIP Client ATA 23 3 2 4 Saving the Configuration Changes Once a change is made the user should press the Update button in the Configuration Menu The IP phone will then display the following screen to confirm that the changes have been saved Your configuration changes have been saved They will take effect on next reboot Users are recommended to power cycle the VOIP Client 488 after seeing the
22. t Universal Power adapter 1 2 Hardware Specification The table below lists the hardware specification of VoIP Client ATA Model VoIP Client ATA LAN interface 1xRJ45 10Base T Button 1 LED GREEN amp RED color Universal Input 100 240VAC Power Adaptor Output 5VDC 1200mA UL certified May 30 2005 VoIP Client ATA 3 Di 65mm W imension 93mm D 27mm H Weight Operating 32 1040F Temperature 0 400C re 10 95 uty non condensing Compliance FCC CE C Tick May 30 2005 VoIP Client ATA 2 Basic Operations 2 1 Getting Familiar with the Key Pad and theVoice Prompt VoIP Client ATA has a stored voice prompt menu for quick browsing and simple configuration To enter this voice prompt menu simple pick up the phone and press the button on the VoIP Client ATA or pick up the phone and dial and after The following table shows how to use the voice prompt menu to configure the device Menu Voice Prompt User s Options Main Menu Enter a Menu Option Enter to next option and back to main menu or Dial 01 06 47 86 or 99 Menu option 01 Static IP Mode or Dynamic IP Mode Dial 9 to toggle the selection If user selects Static IP Mode user will need to configure the all IP address information through menu 02 to 05 If user selects Dynamic IP Mode the device will retrie
23. t menu option returns to the main menu 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 digits are accumulated it automatically processes them Key entry cannot be deleted but the phone may prompt error once it is detected 2 2 Placing Phone Calls 2 2 1 Calling phone or extension numbers There are currently two methods to make an extension number call 1 Dial the extension number directly and wait for 4 seconds Default No Key Entry Timeout Or 2 Dial the number directly and press assuming that Use as dial key is selected in the web configuration Other functions available during the call are call waiting flash call transfer and call forwarding 2 2 2 Direct IP calls Direct IP calling allows two phones to talk to each other in an ad hoc fashion without a SIP proxy VoIP calls can be made between two phones if e Both VOIP Client ATA and the other VoIP device i e another VOIP Client ATA or other SIP products have public IP addresses or e Both VOIP Client ATA and the other VoIP device i e another VOIP Client ATA or other SIP produces are on the same LAN using private or public IP addresses or e Both VOIP Client ATA and the other VoIP device i e another VOIP Client ATA or other SIP products can be connected through a router using public or private IP address
24. then dials C s number and then or waits for 4 seconds 3 A can hang up Note Call Feature has to be set to YES A can hold on to the phone and wait for one of the three following behaviors e A quick confirmation tone temporarily using the call waiting indication tone followed by a dial tone This indicates the transfer is successful transferee has received a 200 OK from transfer target At this point A can either hang up or make another call e A quick busy tone followed by a restored call on supported platforms only This means the transferee has received a 4xx response for the INVITE and we will try to recover the call The busy tone is just to indicate to the transferor that the transfer has failed May 30 2005 VoIP Client ATA 7 e Busy tone keeps playing This means we have failed to receive the second NOTIFY from the transferee and decided to time out Note this does not indicate the transfer has been successful nor does it indicate the transfer has failed When transferee is a client that does not support the second NOTIFY such as our own earlier firmware this will be the case In bad network scenarios this could also happen although the transfer may have been completed successfully 2 2 4 Attended Transfer 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 or Hook Flash for old model phones to get a d
25. ts SIP 2 0 RFC 3261 TCP UDP IP RTP RTCP HTTP ICMP ARP RARP DNS DHCP both client and server NTP PPPoE STUN TFTP etc Powerful digital signal processing DSP to ensure superb audio quality advanced adaptive jitter control and packet loss concealment technology Supports various codecs including G 711 PCM a law and u law G 723 1 5 3K 6 3K G 726 40K 32K 24K 16K as well as G 728 G 729 and iLBC Supports Caller ID name display or block Call waiting caller ID Hold Call Waiting Flash Call Transfer Call Forward in band and out of band DTMF Dial Plans etc Supports fax pass through for PCMU and PCMA and T 38 FoIP Fax over IP Supports Silence Suppression VAD Voice Activity Detection CNG Comfort Noise Generation Line Echo Cancellation G 168 and AGC Automatic Gain Control Supports standard encryption and authentication DIGEST using MD5 and MDS sess Supports for Layer 2 802 1Q VLAN 802 1p and Layer 3 QoS ToS DiffServ MPLS Supports automated NAT traversal without manual manipulation of firewall NAT Supports device configuration via built in IVR Web browser or Central configuration files through TFTP or HTTP server Supports firmware upgrade via TFTP or HTTP with encrypted configuration files Supports PSTN pass through able to make and receive VoIP or PSTN calls using same connected analogue phone Ultra compact wallet size and lightweight design great companion for travelers Compact lightweigh
26. use Yes only if proxy supports 484 response Dial Plan Prefix this prefix string 1s added to each dialed number No Key Entry Timeout Use as Dial Key G yo Ei yes if set to Yes will function as the Re Dial key local SIP port 080 default 5060 local RTP port 004 1024 65535 default 5004 Use random port fi No k in seconds default is 4 seconds LG Yes NAT Traversal G No E Yes STUN server is tezi URI or IP port keep alive interval e in seconds default 20 seconds Use NAT IP if specified this IP address is used in SIP SDP message Proxy Require if specified the content will appear in Proxy Require header Firmware Upgrade Di xi TFIP Server 192 168 fi 30 E va Server 102 108 120 Automatic HTTP Upgrade G G MAN 7 No Yes check for upgrade every davs default 7 davs SUBSCRIBE for i No MITT do not send SUBSCRIBE for Message Waiting Indication u Yes send periodical SUBSCRIBE for Message Waiting Indication Offhook Auto Dial User ID extension to dial automatically when offhook May 30 2005 VoIP Client ATA 16 EnableCall Ei No E Yes Gf Yes Call Forwarding amp Call Waiting Disable are supported locally Disable Call p c Waiting No Yes Features Send DTMF U in audio BB viaRTP RFC2833 via sIPINFO DTMF Payload 101 Type Send Flash Event Ei No L Yes Flash will be sent as a DTMF event if set to Yes FXS Impedan
27. use each G723 voice frame contains 30ms of audio Similarly if this field is set to be 2 and if the first vocoder chosen is G729 or G711 or G726 then the ptime value in the SDP message of an INVITE request will be 20ms If the configured voice frames per TX exceeds the maximum allowed value the phone will use and save the maximum allowed value for the corresponding first vocoder 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 Fax Mode T 38 Auto Detect FoIP by default or Pass Through must use codec PCMU PCMA User ID is phone If the VoIP Client ATA has an assigned PSTN telephone number number then this field will be set to Yes Otherwise set it to No If Yes a user phone parameter will be attached to the From header in SIP request SIP Registration This parameter controls whether the IP phone needs to send REGISTER messages to the proxy server The default setting is Yes Unregister On Default is No If set to Yes the SIP user s registration information will be Reboot cleared on reboot Registration This parameter allows the user to specify the time frequency in minutes the Expiration phone will refresh its registration with the specified registrar The default interval is 60 minutes or 1 hour The maximum interval is 65535 min
28. utes about 45 days Early Dial This parameter controls whether the phone will attempt to send an early INVITE each time a key is pressed when a user is dialing a number If set to Yes an INVITE is sent using the dial numbers collected so far Otherwise no INVITE is sent until the Re Dial button is pressed or after about 5 seconds have elapsed if the user forgets to press the Re Dial button The Yes option should be used ONLY if there is a SIP proxy configured and the proxy server supports 484 Incomplete Address responses Otherwise the call will most likely be rejected by the proxy with a 404 Not Found error Please note that this feature is NOT designed to work with and should NOT be enabled for direct IP to IP calling May 30 2005 VoIP Client ATA 19 This value contains the dial plan prefix string typically an ASCII numeric Dial ian Archia string If it is not blank then this string will be used as a prefix to the target URI string in the To header field of an INVITE message No Key Entry Timeout Default is 4 seconds This parameter allows the user to configure the key to be used as the Send or Dial key Once set to Yes pressing this key will immediately Use as Seng Ke trigger the sending of the l dialed string collected so far In this case this key is essentially equivalent to the Re Dial key If set to No this
29. ve all IP address information from DHCP server automatically when user reboots the device 02 IP Address IP address The current WAN IP address is announced Enter 12 digit new IP address if in Static IP Mode 03 Subnet IP address Same as Menu option 02 04 Gateway IP address Same as Menu option 02 05 DNS Server IP address Same as Menu option 02 06 TFTP Server IP address Same as Menu option 02 TFTP server is used to update the firmware of the device 47 Direct IP Calling When entered user will be prompted by dial tone dial the 12 digit IP address to make a direct IP call For details see 4 2 2 Make a Direct IP Call 86 No Voice Messages or Voice Messages Pending If there are voice messages user can dial 9 and dial pre configured phone number to retrieve voice message 99 RESET Dial 9 to confirm the RESET or Enter MAC address to restore factory default setting For detail see section 8 Invalid Entry Automatically return to Main Menu May 30 2005 VoIP Client ATA Notes Once the LED 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 state it will jump to the Direct IP Calling option dial tone plays in this state shifts down to the nex
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