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Release Note
KX-TGP500 Firmware

Version History

22.85

• Changes/Additions:
  1. Add user-agent header to HTTP GET with authentication header.
  2. Expand the HTTP URL length from 127 to 245 characters.
  3. Be able to switch unREGISTER method, all bind request (Expires *) to all bind request (Contact *)
     Configuration parameter :[ADD_EXPIRES_HEADER_n]
     Y(Add Expires header:default) N(Added no Expires header)

• Improvements/Corrections:
  1. Improved that the terminal returns a 481 response to the new INVITE if the Replaces header field matches an early dialog that was not initiated by the terminal.
  2. Improved that the terminal becomes unavailable when receives invalid URL by provisioning.

22.84

Since the process of download may fail, this firmware is not available.
The function updated and changed with this firmware is included in a version 22.85.

22.80

• Changes/Additions:
  
  1. Added the function to enable or disable Call Waiting for each line.
     Configuration parameter :[CW_ENABLE_n]
  2. Added the function to specify the display of access code or country code in phone number.
     Configuration parameter :
          [INTERNATIONAL_ACCESS_CODE]
          [COUNTRY_CALLING_CODE]
          [NATIONAL_ACCESS_CODE]
          [COUNTRY_CALLING_CODE_EX]
  3. Added the function to confirm by the tone when a conference starts.
     Configuration parameter :[CONFIRM_TONE4_ENABLE]
  4. Enabled the remote reboot by NOTIFY(check-sync).

• Improvements/Corrections:
  
  1. Improved that "Synchronize Do Not Disturb and Call Forward" setting is disabled when phone number is changed at the same time.
  2. Improved the case that 4th party cannot be joined on N-Way conference.

22.73

• Additions:
  
  1. Added the function to change information between "FROM" and "P-Asserted-Identity" fields on SIP information for incoming calls.
  2. Supported the DTMF marker bit of RFC4733.
  3. Added the function to become idle state automatically when the party hangs up the phone.

• Improvements/Corrections:
  
  1. Improved the phenomenon that TGP5xx stops sending "REGISTER" several days after.

22.68

• Additions:
  
  1. Added the following languages on the display.
     Polish,Romanian,Czech,Slovak,Bosnian,Portuguese,Swedish,Norwegian,Danish,Finnish
  2. Added the following languages on Web Interface.
     Polish,Romanian,Czech,Slovak,Portuguese
  3. Added the option to display the number of voice mails on the display.
     Added the parameter:VM_COUNT_ENABLE 
     Y(Enable)  N(Disable:Default)
  4. Added the call rejection function by pressing [OFF] key.
  5. Added the parameter to specify the response code to reject receiving calls.
     Parameter:SIP_RESPONSE_CODE_CALL_REJECT
     Value Range:400-699
  6. Added the function to transfer the holding call.
     Added the parameter:HOLD_TRANSFER_OPERATION
     Y(Enable)  N(Disable:Default)
  7. Enabled to send DTMF following "Pause Dial" after the called party answers.
  8. Added the option to ring a local Ring Back Tone(RBT) when 180 is received after the early media is established.
     Added the parameter:RINGTONE_183_180_ENABLE
     Y(Enable)  N(Disable:Default)
  9. Enabled to call back with the number with "+" when redialing for an international call.
  10. Enabled to input "@" at the Authentication ID and Password on the handset.
  11. Added the softkey [CALL] to send the dial immediately for the second dial.
  12. Changed the value range of RTP port.
      RTP_PORT_MIN   value range:1024�59598
      RTP_PORT_MAX  value range:1424�59998
  13. Enabled to overwrite the Web settings by provisioning.
      Added the parameter:MAINTENANCE_WEB_RESET_ON_STARTUP
      Y(Enable)  N(Disable:Default)
      Regarding Web settings possible to overwrite, refer to "4.3.5 Provisioning Settings" in Administrator Guide.
  14. Added the option to send a request when 403 Forbidden reply is received from the server in response to INVITE or SUBSCRIBE.
      Added the parameter:SIP_403_REG_SUB_RTX_n   (n=1 to 8)
      Y:Enable(Send)  N:Disable(Default,Not Send)
  15. Improved that the parameter:SHARED_CALL_ENABLE_n is set to "N" in case of NULL.

22.58

• Additions:
  
  1. Supported DECT repeater (KX-A405).
  2. Added the function to input the user ID and Password for digest authentication from handset.
  3. Improved to hear the ring-back tone during calling to the destination by REFER transfer.
  4. Expanded the digit of SIP/HTTP User Agent from 40 to 64.
  5. Supported TR-069 remote mainenance.
  6. Supported the access code based redirection service.
  7. Made the phone idle when the other party hung up the call.
  8. Changed the schema of the embedded redirection server URL from http to https automatically.
  9. Added the parameter that specifies the range for the delay interval, in minutes, for the random timing to access the provisioning server.
     New parameter:CFG_RESYNC_DURATION
  10. Added the parameter that Specifies whether or not to add Route headers when setting OutBoundProxy.
      New parameter:SIP_ADD_ROUTE_[1-8]
  11. Supported failover feature.
      -Added new parameter:SIP_FOVR_MODE_[1-8]
         Specifies whether INVITE/SUBSCRIBE will also follow the REGISTER Failover result.
      -Added new parameter:SIP_FOVER_DURATION_[1-8]
         Specifies the number of transmission times for the REGISTER method at the Failover destination.

• Improvements/Corrections:
  
  1. Improved the feature to forward to the conference server.
  2. Improved so that the phone shows the shared line status in case of including Comma in call-info message.
  3. Improved the impossibility of some handset function after web settings are reset when repeater mode is on.
     This is the improvement for ver.22.53 of firmware.

22.38

• Additions:
  
  1. Extended the digit of "Line ID" from 24 to 63.
  2. Added the specification that the # key is treated as a regular dialed digit when dialed as or after the second digit.
     [POUND_KEY_DELIMITER_ENABLE]
     (Y(default) :  # is treated as the end of dialing delimiter)
     (N : # is treated as a regular dialed digit)
     
  3. Added DTMF relay by using SIP-INFO method.
  4. DSCP value could be set separately for RTP and RTCP.
  5. Expanded the size of Refere-To header from 255 bytes to 511 bytes.
  6. Supported for some carriers to use the conference server of another domain.

• Corrections:
  
  1. Improved the problem that the call transfer using REFER may be failed.

22.31

• Additions:
  
  1. TGP500 supported VLAN.
  2. Supported a incoming REFER in transfer call function.
  3. Changed the specification not to display "Missed Call" when any other terminal answers to the imcoming call of the shared line.
  4. Supported handling for the transfer by INVITE with Replaces Header.
  5. Changed a call when "hold" is pressed in both sides at the same time to retry and not to disconnect.
  6. Supported the voicemail function although it is not notified the information of the number of Voicemail Message from the server.
     [VOICE_MESSAGE_AVAILABLE]
     (Y: Determines that voice messages exist when"Messages-Waiting: yes" is received with a "Voice-Message" line included)
     (N: Determines that voice messages exist when"Messages-Waiting: yes" is received even without a "Voice-Message" line included.)
  7. Supported the Pre Provisioning by HTTPS, HTTP, TFTP.
  8. Changed the specification of transfer function. If the destination of call transfer answers by voice-mail, back to the original call easily.
  9. Supported MoH (Music on Hold) of server when connected to particular server.
     [HOLD_SOUND_PATH_n] (0: The unit's hold tone is played. 1: MoH is played.)
  10. Supported for Client Certificate

22.20

• First Release Version