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Release Note
KX-UT113/123/133/136 Firmware

Version History

01.302

• Modification:
  1. Fixed an issue where KX-UT1xx drops the call which is put on hold when answering another call.

• Improvement:
  1. Support 128 bytes of SIP Caller-ID.

01.294

• Modification:
  1. There is that the date of the call history is broken. ex.) 0/0/2016

• Improvement:
  1. Performance improvement that reINVITE can be receive at the same time of the call establishment..
  2. Performance improvement that call disconnection doesn't occur with answer to held call.
     RFC3264 compliant.
  3. Adding new function that SIP privacy header is displyed on BLF.

01.285

• Corrections/Improvements:
  
  1. Improved the phenomenon that lost call when receiving a 183 during transfer outgoing.

01.278

• Enhancements/Additions:
  
  1. Added the PnP provisioning using multi-cast SUBSCRIBE.

• Corrections/Improvements:
  
  1. Improved the phenomenon that the terminal sends 488 response to re-INVITE without SDP.
  2. Improved the phenomenon that the status of BLF becomes different from the actual call status.

01.251

• Enhancements/Additions:
  
  1. Display of DN button is controlled by Shared Call Appearance index.
     (DN button needs to be assigned shared line)
  2. Enabled ID/Password on HTTP digest authentication to enter by phone operation.
  3. Not recorded in Incoming call log when the reason of cancel is "Queue poll".
  4. Changed the default value of Parameter :"AUTO_CALL_HOLD" to "Yes" from "No".
  5. Enabled some WEB settings to change by configuration file.
     Parmaeter:"MAINTENANCE_WEB_RESET_ON_STARTUP" is added.
  6. Added the function to delete all call logs at one time.
  7. Caller ID information on LCD is updated when updating it by P-Asserted-Identity on SIP UPDATE method.
  8. "WorkingAfterCall" is not sent automatically when a conversation finished.

• Corrections/Improvements:
  
  1. Improved that "Synchronize Do Not Disturb and Call Forward" setting is disabled when phone number is changed at the same time.
  2. Made confirmation tone disable when HOLD_AND_CALL_ENABLE="N" and CONFIRM_TONE5_ENABLE="N".

01.221

• Additions:
  
  1. Supported the DTMF marker bit of RFC4733.
  2. Added the function to make idle state automatically when the party hangs up the phone.

• Corrections:
  
  1. Improved so that "HOLD" operation during the conference call makes idle state when HOLD_AND_CALL_ENABLE="N"
  2. Made the voice not muted during call-waiting tone.

01.167

• Additions:
  
  1. Added the following five languages.
     -Bosnian, Romanian, Slovene, Serbian, Turkish
  2. Enabled the transfer after holding operation.
  3. Reduced the load of access to the Provisioning server.
  4. Added ACD status (Available / Unavialable / Wrap up).
  5. Supported LLDP.
     Please refer to No.12-017 of technical contents about LLDP specification.
  6. Improved XML application.
  7. Added the fucntion to make idle status after holding a call.

• Corrections:
  
  1. Enabled "," (comma) to use in a string of Call Information.

01.133

• Additions:
  
  1. The phone number of BLF is automatically assigned by BroadSoft Server.
  2. Improved XML application.
     The phone displays the information from XML server when the call is incoming.
  3. Supported the following ringtone pattern of Bellcore in Alert-info header.
     Bellcore-r1,Bellcore-r2,Bellcore-r3,Bellcore-r4 and Bellcore-r5

• Corrections:
  
  1. Corrected the following issue.
     Even if only ringer tone data is changed on phonebook, it is not changed after importing.

01.111

• Additions:
  
  1. Improved XML Application.
  2. Enabled UT1xx in same group to receive the incoming call by same phone number.

01.081

• Additions:
  
  1. Supported the AnnexF and AnnexG for TR-069.
  2. Supported XML application programing interface(API).
     It is possible to develop various application, such as network phonebook, by using this API.
  3. Added the Missed Call Log menu in the Call Log menu.
  4. Added DTMF relay by using SIP-INFO method.
  5. DSCP value can be set separately for RTP and RTCP.
  6. The User can select whether Key Click tone is ring or not from the phone function menu.

• Corrections:
  
  1. The incoming call is not rejected from similar phone number with own number (e.g. from 101 to 1010)
  2. The receiving volume of SP-phone does not go down when the call waiting tone rings.
  3. Changed LCD display word. (Back Light -> Backlight)
  4. The headset mode status keeps by power OFF/ON of UT1xx.series

01.061

• Additions:
  
  1. Supported a incoming REFER in transfer call function.
  2. Supported Client certificate.
  3. Enabled to answer for the click to dial in the TALK status.
  
  UT113/123: The current call places to hold and answers the call of "the click to dial".
  UT133/136: The current call places to hold or be disconnected.
  4. Supported two kinds of Server certificate.
  5. Enabled to receive the packets of TCP protocol.
  6. Expanded the size of Server certificate used by HTTPS from 5 KByte to 20 KByte.

• Corrections:
  
  1. Reduced the comfort Noise of the SP-Phone sending and Off-Hook monitor by adjusting the parameter.
  2. Reduced the hissing noise of the Handset by adjusting the parameter.

01.025

• Additions:
  
  1. Supported XML provisioning.
  2. Supported TR069 TR104/106.
  3. Supported handling for the transfer by INVITE with Replaces Header.
  4. Added the parameter that specifies whether to add the port number to the Request-Line in the initial SIP request.
  
  [SIP_REQURI_PORT_n](Y:Port Number is added. N: Port Number is not added.)
  5. 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.)
  6. Supported the encoding for ID/PW.
  7. Supported the Pre Provisioning by HTTPS, HTTP, TFTP.
  8. 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.)
  9. Changed to be able to return to 3-way conference after holding the 3-way conference as the function of SIP Phone. (UT133/136)
  10. Removed Default setting (Factory Setting and IP Reset) from the setting of User Level.
  11. Changed to selectable Ringtone on ringer setting.(UT133/136)
  12. Changed to short the DTMF length when pressing the dial key on the post dialing status.

• Corrections:
  
  1. Improved that DTMF dial is detected double.
  2. Supported the domain of "CONFERENCE_SERVER_URI@" on INVITE message when the user calls the conference server.
  3. Supported that codec is changed when the user picks up the held call.

01.002

• First Release Version