Panasonic KX-UT113 SIP Phone Firmware 01.025

Additions: - Supported XML provisioning. - Supported TR069 TR104/106. - Supported handling for the transfer by INVITE with Replaces Header. - 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.) - 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.) - Supported the encoding for ID/PW. - Supported the Pre Provisioning by HTTPS, HTTP, TFTP. - 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.) - Changed to be able to return to 3-way conference after holding the 3-way conference as the function of SIP Phone. (UT133/136) - Removed Default setting (Factory Setting and IP Reset) from the setting of User Level. - Changed to selectable Ringtone on ringer setting.(UT133/136) - Changed to short the DTMF length when pressing the dial key on the post dialing status. Corrections: - Improved that DTMF dial is detected double. - Supported the domain of "CONFERENCE_SERVER_URI@" on INVITE message when the user calls the conference server. - Supported that codec is changed when the user picks up the held call.

  • Разработчик: Panasonic
  • ОС: Windows XP / Windows Vista / Windows XP X64 / Windows Vista64 / Windows 7 / Windows 7 x64
  • Размер: 3.5 MB
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