Features

Ideas

  • Implement 3-way conference or n-way conference using Asterisk bridge
  • Implement support of phone extension panels. Add ability to send several requests in single UDP packet
  • Improve translation by use separate labels for soft keys
  • Improve user expiruence (display missed calls, bitton to access mised calls list)
  • Change mute button function to mute only local mic
  • Save languages/contrast/codecs selected on screen after phone reboot
  • Detect firmware version and decide configuration options automatically
  • Add translations for every language supported
  • Support for ALERT_INFO distinctive ringing
  • Made patch for Unistim support in FreePBX
  • Do away with BUFFSEND
  • Moving phone interface control out into a separate file

Ideas will be implemented on free time available or donations on specific feature available.

Nortel features

  • When a call is on hold, the phone LED BAR blinks
  • There is a part of all Nortel systems called, “held reminder” – set from off, or 30 to 180 seconds. If any held
    line passes the delay time, the phone will “DOOT DOOT” medium volume to remind you someone is waiting.

 

Known issues

  • First, look issue tracker for current known issues
  • Channel not correctly works on PowerPC

Patch#1 functionality (12 March 2012)

  • [option] Added “debug” global option in unistim.conf, that enable debug when module loaded
  • [option] Added “sharpdial” option, enable sending call whet # key pressed
  • [option] Added “cwvolume” and “cwstyle” for callwaiting ring controll
  • [feature] ability for changing display language (tested on Russian language). Use .po files in encoding, able to display ISO 8859-1, ISO 8859-2, ISO 8859-4, ISO 8859-5, ISO 2022-JP. For selecting language can be used option “language” in unistim.conf or screen menu.
  • [feature] Support for multilines Support for holding multiple lines
  • [feature] More fixes for display on i2002 phone
  • [feature] Configurable keys for sending and received history
  • [feature] Menu for selecting codec, contrast (not yet completed) or display language
  • [feature] Show clock at first line of idle phone
  • [feature] Add ability for pick up call Pick up call by using on-screen soft key
  • [feature] Change displaying list of received or send calls (callerid, time and caller name on different screens, listed by lef-right keys)
  • [change] Changed entering on screen phone number, so any number of digits can be entered
  • [change] rtp_port now used start rtp port
  • [change] list of dial tone frequecies now loaded from indications.conf and not hardcoded
  • [change] Key with globe icon how calls menu and not directly codec selection
  • [fix] ASTERISK-16155 Correct updating LED when switching between speekerphone and handset or hanging up
  • [fix] ASTERISK-16087 Multiple crashes when using phone
  • [fix] ASTERISK-15660 Fixed playing dialtone in some scenarious when conversation already started
  • [fix] Fixed dispalying on-screen information when using Redial softkey (DN number and timer displayed).
  • [fix] Not sending short ring in case of call forward enabled on phone
  • Wilfred Blackwell

    Good day,
    I have i2004s installed on an asterisknow 10.13 pbx. I am attempting to perform standard conferencing (not conference bridging). On various sites, some implied unistim supports this such as:
    https://wiki.asterisk.org/wiki/display/AST/Introduction+to+the+Unistim+channel
    which said “This is a channel driver for Unistim protocol. You can use a least a Nortel i2002, i2004 and i2050.
    Following features are supported : Send/Receive CallerID, Redial, SoftKeys, SendText(), Music On Hold, Message Waiting Indication (MWI), Distinctive ring, Transfer, Threeway call, History, Forward, Dynamic SoftKeys.”
    On other sites, its implied that it does not such as :
    https://wiki.asterisk.org/wiki/display/AST/Unistim+channel+improvements
    which said There is still some desired functions missed and please do not look for it in updated chan_unistim:
    3-way call still not implemented
    mute button still mute both parties.
    Can you clear up my confusion? Does the asterisk unistim support regular conferencing. If so, how is it programmed and implemented.
    Found out how to do call pickup after searching the forums:
    “callgroup=1,5-7
    pickupgroup=1”
    but not conferencing. Assistance will be greatly appreciated.

    • There is no built-in conferencing (3-way) in chan_unistim

      • Wilfred Blackwell

        Ok. thanks for the clarification.