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Release Note
KX-TGP500/550 Firmware

Version History

23.35

• Changes/Additions:
  1. Enhance supported number of Call-ID digits. (Up to 128 digits)
  2. Supported Cipher suites of SHA-2 are below.(as same as TGP600)
     �SSL_RSA_WITH_AES_128_CBC_SHA(TLS_RSA_WITH_AES_128_CBC_SHA)
     �SSL_RSA_WITH_3DES_EDE_CBC_SHA(TLS_RSA_WITH_3DES_EDE_CBC_SHA)
     �TLS_RSA_WITH_AES_128_CBC_SHA256(TLS1.2)
     �TLS_RSA_WITH_AES_256_CBC_SHA(TLS1.2)
     �TLS_RSA_WITH_AES_256_CBC_SHA256(TLS1.2)
     
  3. Instoperator password is changed for security reason.

• Improvements/Corrections:
  1. The call is terminated when executing hold of the call at the device.
     After the phone sends INVITE to put someone on HOLD, the phone would cancel the call by
     receiving 200 OK including "m=0".
     RFC 3264 describes that maintaining call when m is 0 also. We conform to this RFC.
     

22.90

• Changes/Additions:
  1. Support new command set for new LCD (new driver chip).
     Old software is not compatible with new LCD module. Therefore, when replacing LCD module with new module at repair, software has to be upraded at the same time.

• Improvements/Corrections:
  1. Collection potentiial problem regarding "call feature menu" selection.

22.88

• Additions:
  1. Supported TR069-Inform SEND by Resync-NOTIFY for TGP500.

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

22.17

• Additions:
  
  1. Support PEM format for certification.
  2. Support VLAN for TGP500.
     This version 22.17 was proven not to support VLAN for TGP500.
  3. Support DHCP option60 with value "Panasonic".
  4. Allow user to access to "Multi Number Settings". [CUSTOM_WEB_PAGE](4:Enable)
  5. Add the setting the HTTP Client Authentication WebGUI for the digest authentication.
  6. Support SIP session refresh by UPDATE method.[SIP_SESSION_METHOD](1:re-INVITE 2:UPDAET 3:auto)
  7. Change SYSLOG output information from "File Error" to "Decord Error", when the decord filure file(configuration/firmware) is detected.
  8. When TGP5xx receives 304 response from the server, TGP acts like TGP succeeded in file download. But TGP runs with the last configuration file. CONFIG_SINGLE_MODE_ENABLE (Y:Enable N:Disable)
  9. It becomes to be possible to downgrade the firmware to version 12.10.
  10. When 181 Call is Being Forward, the incoming ringer is not rung.
  11. Support rport (RFC3581).[SIP_ADD_RPORT_n](Y:Add the rport parameter N:Do not the rport parameter)
  12. Support port punching for RTP.
  13. Remove port paramter on Request URI of initial reqeuest.[SIP_REQURI_PORT_n](Y:Add the port parameter N:Do not the port parameter)
  14. Give up to send REGISTER when receive 403 Forbidden from Server.
  15. SYSLOG output full MAC address.
  16. SYSLOG output with "URL" and "filename".
  17. Change SYSLOG output level from warnning to info, when receive "304 Not Modified" response from sever.
  18. A default value of configuration of DNS failover is changed from "N" to "Y".
  19. A default value of configuration of DNS SRV from "N" to "Y".

• Corrections:
  
  1. Improve the session refresh control for quality.
  2. When the settings from Web GUI reaches more than 256 times, the settings can not be rejected with "Error(Failed)" message on the Web GUI shows.
  3. Change to refresh the session by the result of the negotiation even if the session timer is enable.

22.12

• Additions:
  
  1. The specification of supporting the format of " Messages-Waiting : no " for the notification of voice mail.

• Corrections:
  
  1. Improved that the hold line can not be released in "Line status" menu.

22.10

• Additions:
  
  1. Multi Number Settings can be set up by configuration file.
     Added the following parameter to configuration file.
     INCOMING_CALL_GROUP_[1?8]
     OUTGOING_CALL_LINE_HS[1?6]
     OUTGOING_CALL_LINE_BS
     DEFAULT_LINE_SELECT_HS[1?6]
     DEFAULT_LINE_SELECT_BS
  2. The specification which can select always opening WEB port by configuration file.
     Added parameter : HTTPD_PORTOPEN_AUTO
  3. The automatic reboot setting after the success of pre-provisioning with DHCP server option 66 can be set up by configuration file.
     Added parameter : OPTION66_REBOOT
  4. The specification which can select adding the "Contact" header to SIP ACK message by configuration file.
     Added parameter : SIP_CONTACT_ON_ACK_[1?8]
  5. The specification which can change the volume of outband DTMF tone by configuration file and changed the default value from 0dB to -5dB.
     Added parameter : OUTBANDDTMF_VOL

• Corrections:
  
  1. Improved the display of New Voicemail on TGP550 Base unit.
  2. Improved that the SIP parameter containing a capital letter could not be recognized.
  3. Improved that the voice could not be heard for a few seconds when configuration file was downloaded.

22.04

• Additions:
  
  1. Added the specification of supporting the letters in sip line id for registering SIP server.
     (LINE_ID is added to Configuration file and WEB settings.)
  2. Disclosed the specification of checking the username part of the SIP URI in the "To" header when receiving the INVITE message with an incorrect target SIP URI
     (SIP_RCV_DET_HEADER is added to Configuration, and the default setting of SIP_RCV_DET_HEADER is changed to N from Y.)

• Corrections:
  
  1. Improved handset telephone call sound.
  2. Improved all erase operation of the CID when the group setting of receiving call was changed from the default.

22.02

• First Release Version